SRS-GB28181转RTMP流程

SRS通过SrsSourceManager管理RTMP各种流资源，它管理着SrsSource类，而SrsSource是一个FLV-tag队列，rtmp根据ID查找全局SrsSource对象，获得音视频数据，因此搞懂数据如何在SrsSource中流转，就搞懂了srs如何将GB28181转化为RTMP。

一、GB何时创建SrsSource

1.  媒体服务器向SRS-UDP端口（默认5060）发送register注册信息。

2.  srs收到注册信息后创建SrsGb28181SipSession对象，作为一个与对端交互的会话，随后回复OK。

3.  SrsGb28181SipSession会话定时发送设备查询指令。

4.  对端回复OK

5.  经由3消息，对端返回详细设备列表信息，SrsGb28181SipSession会话收到后，创建与设备ID绑定的channel对象，其中就包含了SrsSource。

6.  回复对端Ok消息

7.  srs发送播放请求，并附带接收端口

8.  对端返回SDP确认信息

9.  对端开始向接收端口发送媒体数据

 

如上分析，可以知道，当媒体服务器返回设备信息列表时【步骤5】，srs根据设备ID创建了全局资源，用于沟通国标协议与RTMP协议。以下是资源创建的关键代码：

srs_error_t SrsGb28181RtmpMuxer::initialize(SrsServer *s, SrsRequest* r)
{
    srs_error_t err = srs_success;

    if (!jitter_buffer) {
        jitter_buffer = new SrsRtpJitterBuffer(channel_id);
    }

    jitter_buffer->SetDecodeErrorMode(kSelectiveErrors);
    jitter_buffer->SetNackMode(kNack, -1, -1);
    jitter_buffer->SetNackSettings(250, 450, 0);

    if (!jitter_buffer_audio) {
        jitter_buffer_audio = new SrsRtpJitterBuffer(channel_id);
    }

    jitter_buffer_audio->SetDecodeErrorMode(kSelectiveErrors);
    jitter_buffer_audio->SetNackMode(kNack, -1, -1);
    jitter_buffer_audio->SetNackSettings(250, 450, 0);

    if (!source_publish) return err;

    req = r;
    server = s;
    
    if ((err = _srs_sources->fetch_or_create(req, (ISrsSourceHandler*)server, &source)) != srs_success) {
        return srs_error_wrap(err, "create source");
    }

    //TODO: ???
    // if (!source->can_publish(false)) {
    //     return srs_error_new(ERROR_GB28181_SESSION_IS_EXIST, "stream %s busy", req->get_stream_url().c_str());
    // }

    if ((err = source->on_publish()) != srs_success) {
        return srs_error_wrap(err, "on publish");
    }

    return err;
}
srs_error_t SrsGb28181RtmpMuxer::initialize(SrsServer *s, SrsRequest* r)
{
    srs_error_t err = srs_success;

    if (!jitter_buffer) {
        jitter_buffer = new SrsRtpJitterBuffer(channel_id);
    }

    jitter_buffer->SetDecodeErrorMode(kSelectiveErrors);
    jitter_buffer->SetNackMode(kNack, -1, -1);
    jitter_buffer->SetNackSettings(250, 450, 0);

    if (!jitter_buffer_audio) {
        jitter_buffer_audio = new SrsRtpJitterBuffer(channel_id);
    }

    jitter_buffer_audio->SetDecodeErrorMode(kSelectiveErrors);
    jitter_buffer_audio->SetNackMode(kNack, -1, -1);
    jitter_buffer_audio->SetNackSettings(250, 450, 0);

    if (!source_publish) return err;

    req = r;
    server = s;
    
    if ((err = _srs_sources->fetch_or_create(req, (ISrsSourceHandler*)server, &source)) != srs_success) {
        return srs_error_wrap(err, "create source");
    }

    //TODO: ???
    // if (!source->can_publish(false)) {
    //     return srs_error_new(ERROR_GB28181_SESSION_IS_EXIST, "stream %s busy", req->get_stream_url().c_str());
    // }

    if ((err = source->on_publish()) != srs_success) {
        return srs_error_wrap(err, "on publish");
    }

 二、接收RTP数据到SrsSource

这一部分会解释SRS如何处理国标RTP数据的，以及如何把处理好的数据转移到SrsSource对象中。

如前图7-9步所示，srs请求实时点播时会附带上接收端口，随后的RTP数据会被发送到此端口上。关于SRS接收RTP的流程，前一节已说明，这里不再赘述，现在看看，SRS通过on_udp_packet

收到RTP数据是如何处理的。

srs_error_t SrsGb28181PsRtpProcessor::on_udp_packet(const sockaddr* from, const int fromlen, char* buf, int nb_buf)
{
    if (config->jitterbuffer_enable){
        return on_rtp_packet_jitter(from, fromlen, buf, nb_buf);
    }else{
        return on_rtp_packet(from, fromlen, buf, nb_buf);
    }
}

on_rtp_packet_jitter与on_rtp_packet都是处理RTP的，功能大同小异，这里只分析on_rtp_packet_jitter。

srs_error_t SrsGb28181PsRtpProcessor::on_rtp_packet_jitter(const sockaddr* from, const int fromlen, char* buf, int nb_buf)
{
    srs_error_t err = srs_success;

    pprint->elapse();
    
    char address_string[64];
    char port_string[16];
    if (getnameinfo(from, fromlen, 
                (char*)&address_string, sizeof(address_string),
                (char*)&port_string, sizeof(port_string),
                NI_NUMERICHOST|NI_NUMERICSERV)){
        // return srs_error_new(ERROR_SYSTEM_IP_INVALID, "bad address");
        srs_warn("gb28181 ps rtp: bad address");
        return srs_success;
    }
    
    int peer_port = atoi(port_string);

    if (true) {
        SrsBuffer stream(buf, nb_buf);
        SrsPsRtpPacket *pkt = new SrsPsRtpPacket();;
        
        if ((err = pkt->decode(&stream)) != srs_success) {
            srs_freep(pkt);
            // return srs_error_wrap(err, "ps rtp decode error");
            srs_warn("gb28181 ps rtp: decode error");
            srs_freep(err);
            return srs_success;
        }
        
        std::stringstream ss3;
        ss3 << pkt->ssrc << ":" << port_string;
        std::string jitter_key = ss3.str();
        
        pkt->completed = pkt->marker;
        
        
        if (pprint->can_print()) {
            srs_trace("<- " SRS_CONSTS_LOG_GB28181_CASTER " gb28181: client_id %s, peer(%s, %d) ps rtp packet %dB, age=%d, vt=%d/%u, sts=%u/%u/%#x, paylod=%dB",
                        channel_id.c_str(),  address_string, peer_port, nb_buf, pprint->age(), pkt->version, 
                        pkt->payload_type, pkt->sequence_number, pkt->timestamp, pkt->ssrc,
                        pkt->payload->length()
                        );
        }
      
        SrsGb28181RtmpMuxer *muxer = fetch_rtmpmuxer(channel_id,  pkt->ssrc);
        if (muxer){
            rtmpmuxer_enqueue_data(muxer, pkt->ssrc, peer_port, address_string, pkt);
        }
     
        SrsAutoFree(SrsPsRtpPacket, pkt);
    }

    return err;
}

on_rtp_packet_jitter做了两件事，

1.解码RTP包

2.将解码后的数据插入缓存。

要说明的是插入的缓存是在全局资源SrsGb28181RtmpMuxer 中的，它就是第一章获取设备时创建的channel，其中包含了关键的SrsSource成员变量，也就是，插入的缓存被处理后，会被全部转移到改成员对象中，等待rtmp请求。所以，我们接下来研究下SrsGb28181RtmpMuxer 。

srs_error_t SrsGb28181RtmpMuxer::do_cycle()
{
    srs_error_t err = srs_success;
    recv_rtp_stream_time =  srs_get_system_time();
    send_rtmp_stream_time = srs_get_system_time();
    uint32_t cur_timestamp = 0;
    int buffer_size = 0;
    bool keyframe = false;
           
    //consume ps stream, and check status
    while (true) {

        pprint->elapse();

        if ((err = trd->pull()) != srs_success) {
            return srs_error_wrap(err, "gb28181 rtmp muxer cycle");
        }

        SrsGb28181Config config = gb28181_manger->get_gb28181_config();

        if (config.jitterbuffer_enable){
            if(jitter_buffer->FoundFrame(cur_timestamp)){
                jitter_buffer->GetFrame(&ps_buffer, ps_buflen, buffer_size, keyframe, cur_timestamp);
            
                if (buffer_size > 0){
                    if ((err = ps_demixer->on_ps_stream(ps_buffer, buffer_size, cur_timestamp, 0)) != srs_success){
                        srs_warn("gb28181: demix ps stream error:%s",  srs_error_desc(err).c_str());
                        srs_freep(err);
                    };
                }
            }
            .......
        }

 

这是SrsGb28181RtmpMuxer 内循环的一小段代码，on_rtp_packet_jitter插入的数据会在这里得到处理，每完整的扒出一段PS帧，就被送入on_ps_stream，此函数会去除PS的同步码，最后通过rtmp_write_packet_by_source将处理后的数据送入SrsSource中。

srs_error_t SrsGb28181RtmpMuxer::rtmp_write_packet_by_source(char type, uint32_t timestamp, char* data, int size)
{
    srs_error_t err = srs_success;

    send_rtmp_stream_time = srs_get_system_time();

    //create a source that will process stream without the need for internal rtmpclient
    if (type == SrsFrameTypeAudio) {
        SrsMessageHeader header;
        header.message_type = RTMP_MSG_AudioMessage;
        // TODO: FIXME: Maybe the tbn is not 90k.
        header.timestamp = timestamp & 0x3fffffff;

        SrsCommonMessage* shared_video = new SrsCommonMessage();
        SrsAutoFree(SrsCommonMessage, shared_video);

        // TODO: FIXME: Check error.
        shared_video->create(&header, data, size);
        source->on_audio(shared_video);
    }else if(type == SrsFrameTypeVideo) {
        SrsMessageHeader header;
        header.message_type = RTMP_MSG_VideoMessage;
        // TODO: FIXME: Maybe the tbn is not 90k.
        header.timestamp = timestamp & 0x3fffffff;

        SrsCommonMessage* shared_video = new SrsCommonMessage();
        SrsAutoFree(SrsCommonMessage, shared_video);

        // TODO: FIXME: Check error.
        shared_video->create(&header, data, size);
        source->on_video(shared_video);
    }
    
    return err;
}