Yearly Archives: 2019

音视频

基于Gstreamer的rtp转rtmp代码

rtp2rtmp_video.c
========

1. 推流到rtmp服务器

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$ ffmpeg -re -i ./fulankelin-hd.mp4 -an -vcodec h264 -f rtp rtp://127.0.0.1:5004 -vn -acodec libopus -f rtp rtp://127.0.0.1:5003
$

2. SDP信息

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SDP:
 
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
t=0 0
a=tool:libavformat 58.29.100
m=video 5004 RTP/AVP 96
c=IN IP4 127.0.0.1
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1
m=audio 5003 RTP/AVP 97
c=IN IP4 127.0.0.1
b=AS:96
a=rtpmap:97 opus/48000/2
a=fmtp:97 sprop-stereo=1
 
.

3. gst-launch测试

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gst-launch-1.0 -em \
  rtpbin name=rtpbin latency=5 \
  udpsrc port=5003 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS" ! rtpbin.recv_rtp_sink_0 \
    rtpbin.  ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! avenc_aac ! mux. \
  udpsrc port=5004 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" ! rtpbin.recv_rtp_sink_1 \
    rtpbin.  ! rtph264depay ! h264parse ! mux. \
  flvmux name=mux streamable=true ! rtmpsink sync=false location=rtmp://u1802/live/demo
 
.

4. 程序代码

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#include <string.h>
#include <math.h>
 
#include <gst/gst.h>
 
#define VIDEO_CAPS "application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"
#define AUDIO_CAPS "application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS"
 
/* will be called when rtpbin has validated a payload that we can depayload */
static void
pad_added_cb(GstElement *rtpbin, GstPad *new_pad, GstElement *depay)
{
    char *pad_name = GST_PAD_NAME(new_pad);
    char *depay_name = gst_element_get_name(depay);
    if (strstr(pad_name, "recv_rtp_src_0_") && strstr(depay_name, "audiodepay"))
    {
        GstPad *sinkpad;
        GstPadLinkReturn lres;
 
        g_print("new payload on rtpbin: %s %s %s\n",
                gst_element_get_name(rtpbin), GST_PAD_NAME(new_pad), gst_element_get_name(depay));
 
        sinkpad = gst_element_get_static_pad(depay, "sink");
        g_assert(sinkpad);
 
        lres = gst_pad_link(new_pad, sinkpad);
        g_assert(lres == GST_PAD_LINK_OK);
        gst_object_unref(sinkpad);
    }
    else if (strstr(pad_name, "recv_rtp_src_1_") && strstr(depay_name, "videodepay"))
    {
        GstPad *sinkpad;
        GstPadLinkReturn lres;
 
        g_print("new payload on rtpbin: %s %s %s\n",
                gst_element_get_name(rtpbin), GST_PAD_NAME(new_pad), gst_element_get_name(depay));
 
        sinkpad = gst_element_get_static_pad(depay, "sink");
        g_assert(sinkpad);
 
        lres = gst_pad_link(new_pad, sinkpad);
        g_assert(lres == GST_PAD_LINK_OK);
        gst_object_unref(sinkpad);
    }
}
 
int main(int argc, char *argv[])
{
    GMainLoop *loop;
    GstElement *pipeline;
 
    GstElement *rtpbin;
    GstElement *audiosrc, *audiodepay, *audiodec, *audiores, *audioconv, *audiosink;
    GstElement *videosrc, *videodepay, *videosink;
    GstElement *flvmux, *rtmpsink;
 
    gboolean res;
    GstCaps *caps;
    GstPadLinkReturn lres;
    GstPad *srcpad, *audio_sinkpad, *video_sinkpad;
 
    gst_init(&argc, &argv);
    pipeline = gst_pipeline_new(NULL);
    g_assert(pipeline);
 
    /* the rtpbin element */
    rtpbin = gst_element_factory_make("rtpbin", "rtpbin");
    g_assert(rtpbin);
    gst_bin_add(GST_BIN(pipeline), rtpbin);
    // 001 源
    audiosrc = gst_element_factory_make("udpsrc", "audiosrc");
    g_assert(audiosrc);
    g_object_set(audiosrc, "port", 5003, NULL);
    caps = gst_caps_from_string(AUDIO_CAPS);
    g_object_set(audiosrc, "caps", caps, NULL);
    gst_caps_unref(caps);
    gst_bin_add(GST_BIN(pipeline), audiosrc);
 
    videosrc = gst_element_factory_make("udpsrc", "videosrc");
    g_assert(videosrc);
    g_object_set(videosrc, "port", 5004, NULL);
    caps = gst_caps_from_string(VIDEO_CAPS);
    g_object_set(videosrc, "caps", caps, NULL);
    gst_caps_unref(caps);
    gst_bin_add(GST_BIN(pipeline), videosrc);
 
    /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
    srcpad = gst_element_get_static_pad(audiosrc, "src");
    audio_sinkpad = gst_element_get_request_pad(rtpbin, "recv_rtp_sink_0");
    lres = gst_pad_link(srcpad, audio_sinkpad);
    g_assert(lres == GST_PAD_LINK_OK);
    gst_object_unref(srcpad);
 
    srcpad = gst_element_get_static_pad(videosrc, "src");
    video_sinkpad = gst_element_get_request_pad(rtpbin, "recv_rtp_sink_1");
    lres = gst_pad_link(srcpad, video_sinkpad);
    g_assert(lres == GST_PAD_LINK_OK);
    gst_object_unref(srcpad);
 
    /* the depayloading and decoding */
    audiodepay = gst_element_factory_make("rtpopusdepay", "audiodepay");
    g_assert(audiodepay);
    audiodec = gst_element_factory_make("opusdec", "audiodec");
    g_assert(audiodepay);
    /* the audio playback and format conversion */
    audioconv = gst_element_factory_make("audioconvert", "audioconv");
    g_assert(audioconv);
    audiores = gst_element_factory_make("audioresample", "audiores");
    g_assert(audiores);
    audiosink = gst_element_factory_make("avenc_aac", "audiosink"); // autoaudiosink voaacenc avenc_aac avenc_opus
    g_assert(audiosink);
    /* add depayloading and playback to the pipeline and link */
    gst_bin_add_many(GST_BIN(pipeline), audiodepay, audiodec, audioconv,
                     audiores, audiosink, NULL);
    res = gst_element_link_many(audiodepay, audiodec, audioconv, audiores,
                                audiosink, NULL);
    g_assert(res == TRUE);
 
    videodepay = gst_element_factory_make("rtph264depay", "videodepay");
    g_assert(videodepay);
    videosink = gst_element_factory_make("h264parse", "videosink");
    g_assert(videosink);
    gst_bin_add_many(GST_BIN(pipeline), videodepay, videosink, NULL);
    res = gst_element_link_many(videodepay, videosink, NULL);
    g_assert(res == TRUE);
 
    // flvmux
    flvmux = gst_element_factory_make("flvmux", "flvmux");
    g_assert(flvmux);
    g_object_set(flvmux, "streamable", TRUE, NULL);
    gst_bin_add(GST_BIN(pipeline), flvmux);
 
    res = gst_element_link(audiosink, flvmux);
    g_assert(res == TRUE);
    res = gst_element_link(videosink, flvmux);
    g_assert(res == TRUE);
 
    rtmpsink = gst_element_factory_make("rtmpsink", "rtmpsink");
    g_assert(rtmpsink);
    g_object_set(rtmpsink, "sync", FALSE, NULL);
    g_object_set(rtmpsink, "location", "rtmp://u1802/live/demo2", NULL);
    gst_bin_add(GST_BIN(pipeline), rtmpsink);
    res = gst_element_link(flvmux, rtmpsink);
    g_assert(res == TRUE);
 
    /* the RTP pad that we have to connect to the depayloader will be created
   * dynamically so we connect to the pad-added signal, pass the depayloader as
   * user_data so that we can link to it. */
    g_signal_connect(rtpbin, "pad-added", G_CALLBACK(pad_added_cb), audiodepay);
    g_signal_connect(rtpbin, "pad-added", G_CALLBACK(pad_added_cb), videodepay);
 
    /* set the pipeline to playing */
    g_print("starting receiver pipeline\n");
    gst_element_set_state(pipeline, GST_STATE_PLAYING);
 
    /* we need to run a GLib main loop to get the messages */
    loop = g_main_loop_new(NULL, FALSE);
    g_main_loop_run(loop);
 
    g_print("stopping receiver pipeline\n");
    gst_element_set_state(pipeline, GST_STATE_NULL);
 
    gst_object_unref(loop);
    gst_object_unref(pipeline);
    gst_object_unref(audio_sinkpad);
    gst_object_unref(video_sinkpad);
    gst_object_unref(rtmpsink);
    gst_object_unref(flvmux);
    gst_object_unref(rtpbin);
    gst_object_unref(audiosrc);
    gst_object_unref(audiodepay);
    gst_object_unref(audiodec);
    gst_object_unref(audiores);
    gst_object_unref(audioconv);
    gst_object_unref(audiosink);
    gst_object_unref(videosrc);
    gst_object_unref(videodepay);
    gst_object_unref(videosink);
    return 0;
}
 
//
音视频

海康威视摄像头

RTSP流格式

rtsp://[username]:[password]@[ip]:[port]/[codec]/[channel]/[subtype]/av_stream

说明:

参数 说明 示例
username 用户名 如admin
password 密码 123456
ip 设备IP 192.168.1.1
port 端口号,默认为554,不填写默认 554
codec 编码 h264,MPEG-4,mpeg4等
channel 通道号,起始为1,通道1则为ch1 ch1
subtype 码流类型,主码流为main,辅码流为sub main

推流

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# ffmpeg -i "rtsp://username:passport@192.168.1.1:554/h264/ch1/sub/av_stream" -vcodec copy -preset:v ultrafast -tune:v zerolatency -acodec copy -f flv  -an "rtmp://192.168.1.1/live/haikang_01"
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with Apple clang version 11.0.0 (clang-1100.0.33.8)
音视频

流媒体服务器nginx-rtmp安装

下载源代码

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# git clone git@github.com:arut/nginx-rtmp-module.git
git clone git@github.com:winshining/nginx-http-flv-module.git
axel -n 100 http://nginx.org/download/nginx-1.17.5.tar.gz

安装依赖

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## ubintu
sudo apt-get install openssl libssl-dev
sudo apt-get install libpcre3 libpcre3-dev
sudo apt-get install zlib1g-dev
 
## centos
sudo yum install -y pcre pcre-devel
sudo yum install -y openssl openssl-devel
sudo yum install -y zlib-devel zlib

编译安装

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tar zxvf nginx-1.17.5.tar.gz
cd nginx-1.17.5/
 ./configure --prefix=/usr/local/nginx --add-module=/home/work/nginx-http-flv-module --with-http_ssl_module --with-debug

配置nginx用户

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sudo useradd nginx
## sudo vim /etc/passwd
## nginx:x:1001:1001:,,,:/home/nginx:/usr/sbin/nologin

创建相关用户

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# 创建相关目录并修改所有者
sudo mkdir -p /usr/local/nginx/data/dash/live
sudo mkdir -p /usr/local/nginx/data/hls/live
sudo mkdir -p /usr/local/nginx/stat
sudo cp /home/work/nginx-http-flv-module/stat.xsl /usr/local/nginx/stat/
 
sudo chown -R nginx /usr/local/nginx/data
sudo chown -R nginx /usr/local/nginx/stat

修改配置文件

见附录:示例

启动nginx服务器

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 .
 
sudo /usr/local/nginx/sbin/nginx

测试

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.
 
# 推流
ffmpeg -re -i /Applications/ambari-vagrant/ubuntu18.4/data/fulankelin-hd.mp4 -c copy -f flv rtmp://u1802/live/fulankelin-hd
 
# 支持播放地址
rtmp://u1802/live/fulankelin-hd
 
http://u1802/live?app=live&stream=fulankelin-hd
http://u1802/live?port=1935&app=live&stream=fulankelin-hd
http://u1802/live/fulankelin-hd.flv
http://u1802/live/fulankelin-hd.flv?port=1935
 
http://u1802/live/fulankelin-hd.mpd
http://u1802/live/fulankelin-hd.m3u8
 
.

配置示例

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user  nginx;
worker_processes  1;
 
 
# error_log  logs/error.log;
# error_log  logs/error.log  notice;
error_log  logs/error.log  debug;
 
pid        logs/nginx.pid;
 
 
events {
    worker_connections  4096;
}
 
http {
    include       mime.types;
    default_type  application/octet-stream;
 
    log_format  main  '$remote_addr - $remote_user [$time_local] "$request" '
                      '$status $body_bytes_sent "$http_referer" '
                      '"$http_user_agent" "$http_x_forwarded_for"';
 
    access_log  logs/access.log  main;
 
    sendfile        on;
    #tcp_nopush     on;
 
    #keepalive_timeout  0;
    keepalive_timeout  65;
 
    #gzip  on;
 
    server {
        listen       80;
        server_name  localhost;
 
        #charset koi8-r;
 
        #access_log  logs/host.access.log  main;
 
        location / {
            root   html;
            index  index.html index.htm;
        }
 
        #error_page  404              /404.html;
 
        # redirect server error pages to the static page /50x.html
        #
        error_page   500 502 503 504  /50x.html;
        location = /50x.html {
            root   html;
        }
 
        # location ~* \.(m3u8)$ {
        #     types {
        #         application/vnd.apple.mpegurl m3u8;
        #         video/mp2t ts;
        #     }
 
        #     root /usr/local/nginx/data;
        #     add_header 'Cache-Control' 'no-cache';
        # }
        location /live {
            flv_live on; # 打开http播放flv直播流的方式
            chunked_transfer_encoding on; # 支持Transfer-Encoding: chunked方式回复
 
            add_header 'Access-Control-Allow-Origin' '*';
            add_header 'Access-Control-Allow-Credentials' 'true';
        }
 
        location ~ \.(mpd|m4a|m4v)$ {
            root /usr/local/nginx/data/dash/;
            add_header 'Cache-Control' 'no-cache';
        }
        # }
        location ~ \.(m3u8|ts)$ {
            types {
                application/vnd.apple.mpegurl m3u8;
                video/mp2t ts;
            }
 
            root /usr/local/nginx/data/hls/;
            add_header 'Cache-Control' 'no-cache';
        }
 
        location ~ \.(flv)$ {
            rewrite ^/(.*)/(.*)\.(flv)$ /$1?app=$1&stream=$2 last;
        }
 
        location /stat {
            rtmp_stat all;
            rtmp_stat_stylesheet stat.xsl;
        }
        location /stat.xsl {
            root /usr/local/nginx/stat/;
        }
 
        # proxy the PHP scripts to Apache listening on 127.0.0.1:80
        #
        #location ~ \.php$ {
        #    proxy_pass   http://127.0.0.1;
        #}
 
        # pass the PHP scripts to FastCGI server listening on 127.0.0.1:9000
        #
        #location ~ \.php$ {
        #    root           html;
        #    fastcgi_pass   127.0.0.1:9000;
        #    fastcgi_index  index.php;
        #    fastcgi_param  SCRIPT_FILENAME  /scripts$fastcgi_script_name;
        #    include        fastcgi_params;
        #}
 
        # deny access to .htaccess files, if Apache's document root
        # concurs with nginx's one
        #
        #location ~ /\.ht {
        #    deny  all;
        #}
    }
 
 
    # another virtual host using mix of IP-, name-, and port-based configuration
    #
    #server {
    #    listen       8000;
    #    listen       somename:8080;
    #    server_name  somename  alias  another.alias;
 
    #    location / {
    #        root   html;
    #        index  index.html index.htm;
    #    }
    #}
 
 
    # HTTPS server
    #
    #server {
    #    listen       443 ssl;
    #    server_name  localhost;
 
    #    ssl_certificate      cert.pem;
    #    ssl_certificate_key  cert.key;
 
    #    ssl_session_cache    shared:SSL:1m;
    #    ssl_session_timeout  5m;
 
    #    ssl_ciphers  HIGH:!aNULL:!MD5;
    #    ssl_prefer_server_ciphers  on;
 
    #    location / {
    #        root   html;
    #        index  index.html index.htm;
    #    }
    #}
 
}
 
rtmp_auto_push on;
rtmp_auto_push_reconnect 1s;
rtmp_socket_dir /tmp;
 
rtmp {
    out_queue           4096;
    out_cork            8;
    max_streams         128;
    timeout             15s;
    drop_idle_publisher 15s;
 
    log_interval 5s; #log模块在access.log中记录日志的间隔时间,对调试非常有用
    log_size     1m; #log模块用来记录日志的缓冲区大小
 
    server {
        listen 1935;
        on_connect http://127.0.0.1:3000/on_connect;
 
        application live {
            live on;
            hls on;
            hls_path /usr/local/nginx/data/hls/live;
            dash on;
            dash_path /usr/local/nginx/data/dash/live;
            gop_cache on; #打开GOP缓存,减少首屏等待时间
 
            notify_update_timeout 30s;
            notify_relay_redirect off; # 启用本地流重定向on_play和on_publish远程重定向。新的流名称是用于远程重定向的RTMP URL的MD5哈希。默认为关闭。
            notify_update_strict off; # 切换on_update回调的严格模式。默认为关闭。打开所有连接错误后,超时以及HTTP解析错误和空响应均被视为更新失败并导致连接终止。
            notify_method get;
 
            on_play http://127.0.0.1:3000/on_play;
            on_publish http://127.0.0.1:3000/on_publish;
            on_done http://127.0.0.1:3000/on_done;
            on_play_done http://127.0.0.1:3000/on_play_done;
            on_publish_done http://127.0.0.1:3000/on_publish_done;
            on_record_done http://127.0.0.1:3000/on_record_done;
            on_update http://127.0.0.1:3000/on_update;
 
        }
    }
}

问题处理

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启动报错
nginx: [warn] 4096 worker_connections exceed open file resource limit: 1024
 
# ulimit -n 65535
 
vim /etc/security/limits.conf
* soft nofile 65535
* hard nofile 65535
 
vim /etc/sysctl.conf 
fs.file-max = 6553560