SRS webrtc 播放源码分析

在上一节中，我们知道在SrsRtcServer::listen_api()中启动了HTTP接口/rtc/v1/play/用于接收RTC播放的请求。

RTC play接口分析

在文件trunk/src/app/srs_app_rtc_api.cpp中。
以http://127.0.0.1:8080/players/rtc_player.html中的RTC播放demo为例。http://127.0.0.1:1985/rtc/v1/play/请求参数为：

api: "http://127.0.0.1:1985/rtc/v1/play/"
clientip: null
sdp: "v=0\r\no=- 3595237597200846526 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE 0 1\r\na=extmap-allow-mixed\r\na=msid-semantic: WMS\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:GDLO\r\na=ice-pwd:XfX/Cs0oIF6lTtQ7/f0oRwKa\r\na=ice-options:trickle\r\na=fingerprint:sha-256 43:F3:3F:99:EF:F0:F7:B0:16:C9:55:0B:48:21:B5:46:E0:63:47:B7:B2:61:70:AE:C1:BC:64:A1:4A:AF:47:2D\r\na=setup:actpass\r\na=mid:0\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 122 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116 37\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:GDLO\r\na=ice-pwd:XfX/Cs0oIF6lTtQ7/f0oRwKa\r\na=ice-options:trickle\r\na=fingerprint:sha-256 43:F3:3F:99:EF:F0:F7:B0:16:C9:55:0B:48:21:B5:46:E0:63:47:B7:B2:61:70:AE:C1:BC:64:A1:4A:AF:47:2D\r\na=setup:actpass\r\na=mid:1\r\na=extmap:14 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:13 urn:3gpp:video-orientation\r\na=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space\r\na=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid\r\na=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id\r\na=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id\r\na=recvonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=fmtp:98 profile-id=0\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 VP9/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 profile-id=2\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:122 VP9/90000\r\na=rtcp-fb:122 goog-remb\r\na=rtcp-fb:122 transport-cc\r\na=rtcp-fb:122 ccm fir\r\na=rtcp-fb:122 nack\r\na=rtcp-fb:122 nack pli\r\na=fmtp:122 profile-id=1\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f\r\na=rtpmap:121 rtx/90000\r\na=fmtp:121 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f\r\na=rtpmap:120 rtx/90000\r\na=fmtp:120 apt=127\r\na=rtpmap:125 H264/90000\r\na=rtcp-fb:125 goog-remb\r\na=rtcp-fb:125 transport-cc\r\na=rtcp-fb:125 ccm fir\r\na=rtcp-fb:125 nack\r\na=rtcp-fb:125 nack pli\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 H264/90000\r\na=rtcp-fb:108 goog-remb\r\na=rtcp-fb:108 transport-cc\r\na=rtcp-fb:108 ccm fir\r\na=rtcp-fb:108 nack\r\na=rtcp-fb:108 nack pli\r\na=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f\r\na=rtpmap:109 rtx/90000\r\na=fmtp:109 apt=108\r\na=rtpmap:35 AV1X/90000\r\na=rtcp-fb:35 goog-remb\r\na=rtcp-fb:35 transport-cc\r\na=rtcp-fb:35 ccm fir\r\na=rtcp-fb:35 nack\r\na=rtcp-fb:35 nack pli\r\na=rtpmap:36 rtx/90000\r\na=fmtp:36 apt=35\r\na=rtpmap:124 H264/90000\r\na=rtcp-fb:124 goog-remb\r\na=rtcp-fb:124 transport-cc\r\na=rtcp-fb:124 ccm fir\r\na=rtcp-fb:124 nack\r\na=rtcp-fb:124 nack pli\r\na=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032\r\na=rtpmap:119 rtx/90000\r\na=fmtp:119 apt=124\r\na=rtpmap:123 H264/90000\r\na=rtcp-fb:123 goog-remb\r\na=rtcp-fb:123 transport-cc\r\na=rtcp-fb:123 ccm fir\r\na=rtcp-fb:123 nack\r\na=rtcp-fb:123 nack pli\r\na=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032\r\na=rtpmap:118 rtx/90000\r\na=fmtp:118 apt=123\r\na=rtpmap:114 red/90000\r\na=rtpmap:115 rtx/90000\r\na=fmtp:115 apt=114\r\na=rtpmap:116 ulpfec/90000\r\na=rtpmap:37 flexfec-03/90000\r\na=rtcp-fb:37 goog-remb\r\na=rtcp-fb:37 transport-cc\r\na=fmtp:37 repair-window=10000000\r\n"
streamurl: "webrtc://127.0.0.1/live/livestream"
tid: "2f62c3e"

我们分析SrsGoApiRtcPlay::do_serve_http是如何进行处理的。
1、对于RTC会话，使用HTTP短连接。

    // For each RTC session, we use short-term HTTP connection.
    SrsHttpHeader* hdr = w->header();
    hdr->set("Connection", "Close");

2、从body中解析请求的json对象

    // Parse req, the request json object, from body.
    SrsJsonObject* req = NULL;
    SrsAutoFree(SrsJsonObject, req);
    if (true) {
        string req_json;
        if ((err = r->body_read_all(req_json)) != srs_success) {
            return srs_error_wrap(err, "read body");
        }

        SrsJsonAny* json = SrsJsonAny::loads(req_json);
        if (!json || !json->is_object()) {
            return srs_error_new(ERROR_RTC_API_BODY, "invalid body %s", req_json.c_str());
        }

        req = json->to_object();
    }

3、从请求对象中获取参数，包括远程对端SDP信息、流地址、客户端IP、api、tid信息。

// Fetch params from req object.
    SrsJsonAny* prop = NULL;
    if ((prop = req->ensure_property_string("sdp")) == NULL) {
        return srs_error_wrap(err, "not sdp");
    }
    string remote_sdp_str = prop->to_str();

    if ((prop = req->ensure_property_string("streamurl")) == NULL) {
        return srs_error_wrap(err, "not streamurl");
    }
    string streamurl = prop->to_str();

    string clientip;
    if ((prop = req->ensure_property_string("clientip")) != NULL) {
        clientip = prop->to_str();
    }

    string api;
    if ((prop = req->ensure_property_string("api")) != NULL) {
        api = prop->to_str();
    }

    string tid;
    if ((prop = req->ensure_property_string("tid")) != NULL) {
        tid = prop->to_str();
    }

4、解析出application和流名称。和RTMP中的类似。

    // TODO: FIXME: Parse vhost.
    // Parse app and stream from streamurl.
    string app;
    string stream_name;
    if (true) {
        string tcUrl;
        srs_parse_rtmp_url(streamurl, tcUrl, stream_name);

        int port;
        string schema, host, vhost, param;
        srs_discovery_tc_url(tcUrl, schema, host, vhost, app, stream_name, port, param);
    }

5、客户端指定信息

   // For client to specifies the EIP of server.客户端指定服务器公网IP
    string eip = r->query_get("eip");
    string codec = r->query_get("codec");
    // For client to specifies whether encrypt by SRTP.客户端指定是否使用SRTP加密
    string srtp = r->query_get("encrypt");
    string dtls = r->query_get("dtls");

6、创建RTC用户配置对象，并设置相应的参数值。

    SrsRtcUserConfig ruc;
    ruc.eip_ = eip;
    ruc.codec_ = codec;
    ruc.publish_ = false;
    ruc.dtls_ = (dtls != "false");

    if (srtp.empty()) {
        ruc.srtp_ = _srs_config->get_rtc_server_encrypt();
    } else {
        ruc.srtp_ = (srtp != "false");
    }
    
    // TODO: FIXME: It seems remote_sdp doesn't represents the full SDP information.
    // 解析远端的SDP信息，即解析HTTP请求body中的sdp信息
    if ((err = ruc.remote_sdp_.parse(remote_sdp_str)) != srs_success) {
        return srs_error_wrap(err, "parse sdp failed: %s", remote_sdp_str.c_str());
    }

    if ((err = check_remote_sdp(ruc.remote_sdp_)) != srs_success) {
        return srs_error_wrap(err, "remote sdp check failed");
    }
    //设置application和流名称
    ruc.req_->app = app;
    ruc.req_->stream = stream_name;
    // TODO: FIXME: Parse vhost.
    // discovery vhost, resolve the vhost from config
    SrsConfDirective* parsed_vhost = _srs_config->get_vhost("");
    if (parsed_vhost) {
        ruc.req_->vhost = parsed_vhost->arg0();
    }
    //设置本地的SDP信息
    SrsSdp local_sdp;

    // Config for SDP and session.
    local_sdp.session_config_.dtls_role = _srs_config->get_rtc_dtls_role(ruc.req_->vhost);
    local_sdp.session_config_.dtls_version = _srs_config->get_rtc_dtls_version(ruc.req_->vhost);
    
    ......
    
     // TODO: FIXME: When server enabled, but vhost disabled, should report error.
     //创建RTC连接会话
    SrsRtcConnection* session = NULL;
    if ((err = server_->create_session(&ruc, local_sdp, &session)) != srs_success) {
        return srs_error_wrap(err, "create session, dtls=%u, srtp=%u, eip=%s", ruc.dtls_, ruc.srtp_, eip.c_str());
    }

    //创建HTTP返回结果
    ostringstream os;
    if ((err = local_sdp.encode(os)) != srs_success) {
        return srs_error_wrap(err, "encode sdp");
    }

    string local_sdp_str = os.str();
    // Filter the \r\n to \\r\\n for JSON.
    local_sdp_str = srs_string_replace(local_sdp_str.c_str(), "\r\n", "\\r\\n");

    res->set("code", SrsJsonAny::integer(ERROR_SUCCESS));
    res->set("server", SrsJsonAny::str(SrsStatistic::instance()->server_id().c_str()));

    // TODO: add candidates in response json?

    res->set("sdp", SrsJsonAny::str(local_sdp_str.c_str()));
    res->set("sessionid", SrsJsonAny::str(session->username().c_str()));

SRS的Demo中HTTP返回结果为：

{
    "code": 0,
    "server": "vid-0705009",
    "sdp": "v=0\r\no=SRS/4.0.126(Leo) 4336961024 2 IN IP4 0.0.0.0\r\ns=SRSPlaySession\r\nt=0 0\r\na=ice-lite\r\na=group:BUNDLE 0 1\r\na=msid-semantic: WMS live/livestream\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:92994v15\r\na=ice-pwd:x5b70262a62515ab7931y88dv6sc4a02\r\na=fingerprint:sha-256 F2:F0:91:12:9E:EE:24:F7:4C:BA:BA:76:69:C7:31:06:47:46:08:99:7A:53:B3:3B:24:4B:0D:2C:F8:86:76:49\r\na=setup:passive\r\na=mid:0\r\na=sendonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:111 opus/48000/2\r\na=ssrc:59979745 cname:42o749vdf7rz0i14\r\na=ssrc:59979745 label:audio-7o2g616s\r\na=candidate:0 1 udp 2130706431 10.73.85.53 8000 typ host generation 0\r\nm=video 9 UDP/TLS/RTP/SAVPF 125\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:92994v15\r\na=ice-pwd:x5b70262a62515ab7931y88dv6sc4a02\r\na=fingerprint:sha-256 F2:F0:91:12:9E:EE:24:F7:4C:BA:BA:76:69:C7:31:06:47:46:08:99:7A:53:B3:3B:24:4B:0D:2C:F8:86:76:49\r\na=setup:passive\r\na=mid:1\r\na=sendonly\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:125 H264/90000\r\na=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f\r\na=ssrc:59979746 cname:42o749vdf7rz0i14\r\na=ssrc:59979746 label:video-4l550c0u\r\na=candidate:0 1 udp 2130706431 193.168.53.66 8000 typ host generation 0\r\n",
    "sessionid": "92994v15:G14I"
}

193.168.53.66是在rtc.conf中配置的candidate值，本机的局域网地址。

创建RTC会话

create_session不仅是创建会话，同时还会将local_sdp进行赋值。

SrsRtcServer::create_session

1、根据req获取或创建source，用于_srs_rtc_sources管理。

    SrsRequest* req = ruc->req_;

    SrsRtcSource* source = NULL;
    if ((err = _srs_rtc_sources->fetch_or_create(req, &source)) != srs_success) {
        return srs_error_wrap(err, "create source");
    }

2、创建会话

    // TODO: FIXME: add do_create_session to error process.
    SrsRtcConnection* session = new SrsRtcConnection(this, cid);
    if ((err = do_create_session(ruc, local_sdp, session)) != srs_success) {
        srs_freep(session);
        return srs_error_wrap(err, "create session");
    }

3、首先根据ruc->publish_类型为sdp媒体信息添加发布者/播放者

    // first add publisher/player for negotiate sdp media info
    if (ruc->publish_) {
        if ((err = session->add_publisher(ruc, local_sdp)) != srs_success) {
            return srs_error_wrap(err, "add publisher");
        }
    } else {
        if ((err = session->add_player(ruc, local_sdp)) != srs_success) {
            return srs_error_wrap(err, "add player");
        }
    }

添加播放者

由于本章我们是介绍RTC播放的内容，所以本节选取session->add_player进行分析。

add_player

create_player

如果已经创建过，直接忽略，返回成功。

    // Ignore if exists.
    if(players_.end() != players_.find(req->get_stream_url())) {
        return err;
    }

创建SrsRtcPlayStream player，并初始化player。将player加入到players_的map表中。

    SrsRtcPlayStream* player = new SrsRtcPlayStream(this, _srs_context->get_id());
    if ((err = player->initialize(req, sub_relations)) != srs_success) {
        srs_freep(player);
        return srs_error_wrap(err, "SrsRtcPlayStream init");
    }
    players_.insert(make_pair(req->get_stream_url(), player));

判断SSRC是否是重复的。

for(map<uint32_t, SrsRtcTrackDescription*>::iterator it = sub_relations.begin(); it != sub_relations.end(); ++it) {
}

最后，如果DTLS握手成功，则直接启动player。

    if(ESTABLISHED == state_) {
        if(srs_success != (err = player->start())) {
            return srs_error_wrap(err, "start player");
        }
    }

SrsRtcPlayStream类

启动后，st_thread协程开始执行SrsRtcPlayStream::cycle()，可以和上一节（RTMP推流，RTC播放）中的最后一步接续上。

SrsRtcPlayStream::cycle

1、创建consumer，并将其加入到SrsRtcSource source_的consumers数组中，便于当publisher推流时，将音视频数据循环发送给每个consumers。

    SrsRtcConsumer* consumer = NULL;
    SrsAutoFree(SrsRtcConsumer, consumer);
    if ((err = source->create_consumer(consumer)) != srs_success) {
        return srs_error_wrap(err, "create consumer, source=%s", req_->get_stream_url().c_str());
    }