音视频流媒体开发-目录
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SRS流媒体框架

一、推流

1. 监听完后，是不是有个循环处理accept?因为要获取新的连接
   每个监听都会对应一个协程
   每个客户端连接也会对应一个协程

2. 监听类型

   
   
   

   涉及到创建连接的时候知道什么连接，http 、rtmp要创建不同的连接对象

3. 监听和创建协程

4. fd和connection的绑定一》SrsServer : :fd2conn -> new Sr sRtmpConn(this, stfd, ip)

   
   

SRS流媒体服务器-推流框架分析

核心类:

• SrsServer SRS流媒体服务入口
• SrsBufferListener 监听器，主要是TCP的监听
• SrsTcpListener TCP监听器
• SrsRtmpConn RTMP连接，里面对应了SrsStSocket和SrsCoroutine
• SrsRtmpServer提供与客户端之间的RTMP-命令-协议-消息的交互服务，使用* * * SrsRtmpConn提供的socket读写数据
• SrsSource描述一路播放源，包括推流和拉流的描述
• SrsConsumer拉流消费者，每一路拉流客户端对应一个SrsConsumer
• SrsStSocket经过封装的socket接口

SrsRecvThread负责接收数据，但是要注意的是他这里并不是从IO里面读取数据

从SrsRtmpServer类拉取数据，然后推送到SrsPublishRecvThread(推流用)，或者SrsQueueRecvThread(拉流用)
SrsQueueRecvThread主要用于拉流
SrsPublishRecvThread主要用于推流

在客户端进行推流验证

ffmpeg -re -i rtmp_test_hd.flv -vcodec copy -acodec copy -f flv -y rtmp:/111.229.231.225/live/livestream

下断点

b SrsRtmpConn:publishing(SrsSource*)

Breakpoint 8,SrsRtmpConn:publishing (this=Oxa30d00, source=Oxa3bfc0) atsrc/app/srs_app_rtmp_conn.cpp:806806 {

(gdb) bt

#0SrsRtmpConn:publishing (this=0xa30d00, source=Oxa3bfc0) at src/app/srs_app_rtmp_conn.cpp:806
#1 0x00000000004d5229 in SrsRtmpConn:stream_service_cycle (this=Oxa30d00) at src/app/srs_app_rtmp_conn.cpp:534
#2 0x00000000004d4141 in SrsRtmpConn:service_cycle (this=Oxa30d00) atsrc/app/srs_app_rtmp_conn.cpp:388
#3 Ox00000000004d2f09 in SrsRtmpConn:do_cycle (this=Oxa30d00) at src/app/srs_app_rtmp_conn.cpp:209#4 Ox00000000004d10fb in SrsConnection:cycle (this=Oxa30d78) at stc/app/srs_app_conn.cpp:171
#5 Ox0000000000509c88 in SrsSTCoroutine:cycle (this=Oxa30fb0) at src/app/srs_app_st.cpp:198
#6 0x0000000000509cfd in SrsSTCoroutine:pfn (arg=Oxa30fb0) at src/app/srs_app_st.cpp:213
#7 0x00000000005bdd9d in _st_thread_main () at sched.c:337
#8 0x00000000005be515 in st_thread_create (start=Ox5bd719<_st_vp_schedule+170>, arg=Ox700000001,
jginable=1,

st_netfd_poll

二、SRS流媒体服务器-RTMP拉流框架分析

核心类

• SrsServer SRS流媒体服务入口
• SrsBufferListener监听器，主要是TCP的监听
• SrsTcpListener TCP监听器
• SrsRtmpConn RTMP连接，里面对应了SrsStSocket和SrsCoroutine
• SrsRtmpServer提供与客户端之间的RTMP-命令-协议-消息的交互服务，使用* SrsRtmpConn 提供的socket读写数据
• SrsSource描述一路播放源，包括推流和拉流的描述
• SrsConsumer拉流消费者，每一路拉流客户端对应一个SrsConsumer
• SrsStSocket经过封装的socket接口。

SrsRecvThread负责接收数据，但是要注意的是他这里并不是从IO里面读取数据

从SrsRtmpServer类拉取数据，然后推送到SrsPublishRecvThread(推流用)，或者SrsQueueRecvThread (拉流用)

SrsQueueRecvThread主要用于拉流，对应的是客户端-服务器的控制消息，和音视频消息没有关系。客户端读取数据还是从consumer的queue里面去读取。

SrsPublishRecvThread主要用于推流

测试客户端

在客户端进行推流验证
ffmpeg -re -i rtmp_test_hd.flv -vcodec copy -acodec copy -f flv -y rtmp://111.229.231.225/live/livestream

在客户端拉流验证

ffplay rtmp://111.229.231.225/live/livestream

重点难点

不同协程的意义

打断点

客户端和服务器直接的交互，非音视频数据

断点: b SrsRtmpConn::process_play_control_msg(SrsConsumer* ,SrsCommonMessage*)

打印: print *msg

$3 =ivptr.SrsCommonMessage = 0x6b79a8<vtable for SrsCommonMessage+16> , header = (
_vptr.SrsMessageHeader = Ox6b79d8<vtable for SrsMessageHeader+16> , timestamp_delta = 1,payload_length = 10,
message_type = 4 '\004', stream_id = 0, timestamp = 1, perfer_cid = 2}, size = 10, payload = Oxa3aa80""
$4 = fvptr.SrsCommonMessage = 0x6b79a8 <vtable for SrsCommonMessage+16>, header = {
_ vptr.SrsMessageHeader = Ox6b79d8 <vtable for SrsMessageHeader+16> , timestamp_delta = 9130,payload_length = 4,
message_type = 3 1003', stream_id = 0, timestamp = 9131, perfer_cid = 2), size = 4, payload = Oxa74580""}
$5 = f_vptr.SrsCmmonMessage = 0x6b79a8<vtable for SrsCommonMessage+16> , header = f
_ vptr.SrsMessageHeader = Ox6b79d8<vtable for SrsMessageHeader+16>, timestamp_delta = 9280,payload_length = 4,
message type = 3\003', stream_jid = 0, timestamp = 30731, perfer_cid = 2}), size = 4, payload = 0x10325f0""}

以ffmpeg为例

*
* known RTMP packet types
*/
typedef enum RTMPPacketType {
  RTMP_PT_CHUNK SIZE =1, /// <chunk size change
  RTMP_PT_BYTES_READ = 3, ///< 3 number of bytes read
  RTMP_PT_USER_CONTROL, ///< 4 user control
  RTMP_PT_WINDOW_ACK_SIZE, /// < window acknowledgement size
  RTMP_PT_SET_PEER_BW, ///< peer bandwidth
  RTMP_PT_AUDIO= 8，/// < audio packet
  RTMP_PT_VIDEO, ///<video packet
  RTMP_PT_FLEX_STREAM = 15, /// < Flex shared stream
  RTMP_PT_FLEX_OBJECT,  ///< Flex shared object
  RTMP_PT_FLEX_MESSAGE, /// <Flex shared message
  RTMP_PT_NOTIFY, /// < some notification
  RTMP_PT_SHARED_OBJ, ///< shared object
  RTMP_PT_INVOKE, ///<invoke some stream action
  RTMP_PT_METADATA = 22, /// < FLV metadata
}RTMPPacketType;

客户端读取的包大于>receive_report_size时，回复RTMP_Pr_BYTES_READ
receive_report_size来自RTMP_PT_WINDOW_ACK_SIZE消息ID

rt->bytes_read += ret;
if (rt->bytes_read - rt->last_bytes_read  rt->receive_report_size){
  av_log(s, AV_LOG_DEBUG，"sending bytes read report\n" );
  if ( (ret = gen_bytes_read(s， rt,rpkt.timestamp + 1))< 0) {
    ff_rtmp_packet_destroy (&rpkt);
    return ret;
  }
  rt->last_bytes_read = rt->bytes_read;
}