基于HTTP Live Streaming（HLS） 搭建在线点播系统

1、 为何要使用HTTP Live Streaming

可以参考wikipedia
HTTP Live Streaming（缩写是 HLS）是一个由苹果公司提出的基于HTTP的流媒体 网络传输协议。
是苹果公司QuickTime X和iPhone软件系统的一部分。
它的工作原理是把整个流分成一个个小的基于HTTP的文件来下载，每次只下载一些。
当媒体流正在播放时，客户端可以选择从许多不同的备用源中以不同的速率下载同样的资源，允许流媒体会话适应不同的数据速率。
在开始一个流媒体会话时，客户端会下载一个包含元数据的extended M3U (m3u8) playlist文件，用于寻找可用的媒体流。


HLS只请求基本的HTTP报文，与实时传输协议（RTP)不同，HLS可以穿过任何允许HTTP数据通过的防火墙或者代理服务器。
它也很容易使用内容分发网络来传输媒体流。

2、 HTTP Live Streaming技术方案

HTTP服务：使用Nginx提供HTTP服务，通过使用nginx-rtmp-module https://github.com/arut/nginx-rtmp-module来增加对HLS的支持
使用ffmpeg来完成对flv、mp4、mp3等格式的转化
使用ffmpeg或segmenter完成对视频/音频格式文件的切割，切割为m3u8格式及ts文件

3、 准备工作

操作系统
CentOS
3.1、准备安装删除已安装包
$ yum erase ffmpeg x264 x264-devel
3.2、安装各种依赖包
yum install  gcc make nasm pkgconfig wget curl curl-devel zlib-devel openssl-devel perl cpio expat-devel gettext-devel libtool mhash.x86_64 perl-Digest-SHA1.x86_64  pcre.i386 pcre.x86_64 pcre-devel.i386 pcre-devel.x86_64
3.3、安装git
wget -O git-devel.zip https://github.com/msysgit/git/archive/devel.zip
unzip git-devel.zip
cd git-devel/
autoconf
./configure
make
make install
3.4、创建安装包目录
mkdir ~/ffmpeg-source
​

4、 安装ffmpeg及其依赖包

4.1、Yasm
cd ~/ffmpeg-source
wget http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz
tar xzvf yasm-1.2.0.tar.gz
cd yasm-1.2.
./configure
make
make install


4.2、x264
cd ~/ffmpeg-source
git clone git://git.videolan.org/x264
cd x264
./configure –enable-shared
make
make install


4.3、LAME
cd ~/ffmpeg-source
wget http://downloads.sourceforge.net/project/lame/lame/3.99/lame-3.99.5.tar.gz
tar xzvf lame-3.99.5.tar.gz
cd lame-3.99.5
./configure –enable-nasm
make
make install


4.4、libogg
cd ~/ffmpeg-source
wget http://downloads.xiph.org/releases/ogg/libogg-1.3.0.tar.gz
tar xzvf libogg-1.3.0.tar.gz
cd libogg-1.3.0
./configure
make
make install


4.5、libvorbis
cd ~/ffmpeg-source
wget http://downloads.xiph.org/releases/vorbis/libvorbis-1.3.3.tar.gz
tar xzvf libvorbis-1.3.3.tar.gz
cd libvorbis-1.3.3
./configure
make
make install


4.6、libvpx
cd ~/ffmpeg-source
git clone http://git.chromium.org/webm/libvpx.git
cd libvpx
./configure  –enable-shared
make
make install


4.7、 FAAD2
cd ~/ffmpeg-source
wget http://downloads.sourceforge.net/project/faac/faad2-src/faad2-2.7/faad2-2.7.tar.gz
tar zxvf faad2-2.7.tar.gz
cd faad2-2.7
./configure
make
make install


4.8、FAAC
cd ~/ffmpeg-source
wget http://downloads.sourceforge.net/project/faac/faac-src/faac-1.28/faac-1.28.tar.gz
tar zxvf faac-1.28.tar.gz
cd faac-1.28
./configure
make
make install


4.9、Xvid
cd ~/ffmpeg-source
wget http://downloads.xvid.org/downloads/xvidcore-1.3.2.tar.gz
tar zxvf xvidcore-1.3.2.tar.gz
cd xvidcore/build/generic
./configure
make
make install
4.10、FFmpeg
cd ~/ffmpeg-source
git clone git://source.ffmpeg.org/ffmpeg
cd ffmpeg
./configure  –enable-version3  –enable-libvpx –enable-libfaac –enable-libmp3lame  –enable-libvorbis –enable-libx264 –enable-libxvid –enable-shared –enable-gpl –enable-postproc –enable-nonfree –enable-avfilter –enable-pthreads
make
make install
ldconfig –v

5、 安装nginx及nginx-rtmp-module

mkdir ~/nginx-source
cd  ~/nginx-source
wget http://nginx.org/download/nginx-1.2.4.tar.gz
tar zxvf nginx-1.2.4.tar.gz
wget -O nginx-rtmp-module.zip  https://github.com/arut/nginx-rtmp-module/archive/master.zip
unzip nginx-rtmp-module.zip
wget -O ngx_cache_purge.zip https://github.com/FRiCKLE/ngx_cache_purge/archive/master.zip
unzip ngx_cache_purge.zip
cd nginx-1.2.4
./configure –user=daemon –group=daemon –prefix=/usr/local/nginx-1.2.4/ –add-module=../nginx-rtmp-module-master –add-module=../nginx-rtmp-module-master/hls –add-module=../ngx_cache_purge-master  –with-http_stub_status_module –with-http_ssl_module –with-http_sub_module –with-md5=/usr/lib –with-sha1=/usr/lib –with-http_gzip_static_module
在nginx.conf中增加rtmp模块的相关配置，配置例子
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 32×32. 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 /usr/local/bin/ffmpeg -re -i rtmp://localhost:1935/$app/$name -vcodec flv -acodec copy -s 32×32 -f flv rtmp://localhost:1935/small/${name};
              }
              application small {
                     live on;
                     # Video with reduced resolution comes here from ffmpeg
              }
              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;
              }
              # video on demand
              application vod {
                     play /var/flvs;
              }
              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 /var/vc;
                     record_max_frames 10;
                     record_interval 2m;
                     # Async notify about an flv recorded
                     on_record_done http://localhost:8080/record_done;
              }
              # HLS
              # HLS requires libavformat & should be configured as a separate
              # NGINX module in addition to nginx-rtmp-module:
              # ./configure … –add-module=/path/to/nginx-rtmp-module/hls …
              # For HLS to work please create a directory in tmpfs (/tmp/app here)
              # for the fragments. The directory contents is served via HTTP (see
              # http{} section in config)
              #
              # Incoming stream must be in H264/AAC/MP3. 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 {
                     hls on;
                     hls_path /var/app;
                     hls_fragment 5s;
              }
       }
}


# HTTP can be used for accessing RTMP stats
http {
       server {
              listen      8080;
              # 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 /path/to/stat.xsl/;
              }
              location /hls {
                     # Serve HLS fragments
                     alias /var/app;
              }
       }
}

6、 安装segmenter

svn co http://httpsegmenter.googlecode.com/svn/
cd  svn/trunk
gcc -Wall -g segmenter.c -o segmenter -lavformat -lavcodec -lavutil -std=c99
cp segmenter /usr/bin/
从优酷上下载一个视频文件，假定为baluobu.flv
找个mp3文件，假定为10year.mp3
mkdir /var/flvs /var/mp4s /var/vc /var/app /var/app/10year /var/app/baluobu
使用ffmpeg将测试视频和音频转为mpeg ts格式文件
ffmpeg -i /var/flvs/baluobu.flv  -f mpegts -acodec libmp3lame -ar 48000 -ab 128k -vcodec libx264 -b 96k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 200k -maxrate 96k -bufsize 96k -rc_eq ‘blurCplx^(1-qComp)’ -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect 320:240 -g 30 -async 2 /var/app/baluobu/baluobu.ts
ffmpeg -i /var/flvs/10year.mp3  -f mpegts -acodec libmp3lame -ar 48000 -ab 128k -vcodec libx264 -b 96k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 200k -maxrate 96k -bufsize 96k -rc_eq ‘blurCplx^(1-qComp)’ -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect 320:240 -g 30 -async 2 /var/app/10year/10year.ts
cd /var/app/10year
segmenter -i 10year.ts -d 5 -o 10year -x 10year.m3u8
cd /var/app/baluobu
segmenter -i baluobu.ts -d 5 -o buluobu -x baluobu.m3u8

7、测试

简单起见使用VLC测试。
http://192.168.1.11:8080/hls/10year/10year.m3u8
http://192.168.1.11:8080/hls/baluobu/baluobu.m3u8
如果无问题，说明已经正常配置成功了HTTP Live Streaming（HLS）服务
可以再iOS设备上使用safari访问以上地址
或者在android机器上安装vplayer后访问以上地址
如果需要通过网页形式提供在线视频，需要在nginx的mime.types文件中添加如下MIME类型：
.m3u8 application/x-mpegURL
.ts video/MP2T
同时在HTML5页面中使用video标签包含视频的m3u8文件
<video controls>
<source src=http://192.168.1.11:8080/hls/baluobu/baluobu.m3u8 />
</video>

8、开发说明

iOS已经原生支持HTTP Live Streaming（HLS），只需要使用MPMoviePlayerController播放以上地址即可
apple文档：http://bit.ly/Rnpsef
Android 3.x以后通过新增的NuPlayer类才支持HTTP Live streaming，而且功能也较弱，为简化处理，可以使用
Vitamio     http://vov.io/vitamio/ 或 http://code.taobao.org/p/oplayer/src/ 下载
Servestream http://sourceforge.net/projects/servestream/
nginx-rtmp-module 本身也支持RTMP协议，从而也可以基于nginx来搭建在线直播系统。