Merge pull request #5 from xiongziliang/master

update from origin

ZLToolKit @ 60aab3b9

-Subproject commit ed47015f92cc79dfe3344b3666aafb54f1bbc2f4
+Subproject commit 60aab3b9f6230312c882b9d3d360ed27c94ebd9f

media-server @ 43facc34

-Subproject commit 7f7906b05d84c5efeceecb8d6f540a71c8153431
+Subproject commit 43facc343afc2b5b70bbbc3c177f20dfa936f2bf

postman/127.0.0.1.postman_environment.json

+{
+	"id": "95afe791-f716-426e-99c4-a797e112ab2c",
+	"name": "127.0.0.1",
+	"values": [
+		{
+			"key": "ZLMediaKit_URL",
+			"value": "127.0.0.1",
+			"enabled": true
+		},
+		{
+			"key": "ZLMediaKit_secret",
+			"value": "035c73f7-bb6b-4889-a715-d9eb2d1925cc",
+			"enabled": true
+		},
+		{
+			"key": "defaultVhost",
+			"value": "__defaultVhost__",
+			"enabled": true
+		}
+	],
+	"_postman_variable_scope": "environment",
+	"_postman_exported_at": "2020-07-11T15:16:04.479Z",
+	"_postman_exported_using": "Postman/7.27.1"
+}
\ No newline at end of file

postman/readme.md

+把这两个json文件导入postman就可以愉快的测试ZLMediaKit的restful接口了
\ No newline at end of file

server/WebApi.cpp

@@ -257,7 +257,7 @@ static recursive_mutex s_ffmpegMapMtx;
 
 #if defined(ENABLE_RTPPROXY)
 //rtp服务器列表
-static unordered_map<uint16_t, RtpServer::Ptr> s_rtpServerMap;
+static unordered_map<string, RtpServer::Ptr> s_rtpServerMap;
 static recursive_mutex s_rtpServerMapMtx;
 #endif
 
@@ -753,44 +753,53 @@ void installWebApi() {
         val["exist"] = true;
         val["peer_ip"] = process->get_peer_ip();
         val["peer_port"] = process->get_peer_port();
+        val["local_port"] = process->get_local_port();
+        val["local_ip"] = process->get_local_ip();
     });
 
     api_regist1("/index/api/openRtpServer",[](API_ARGS1){
         CHECK_SECRET();
         CHECK_ARGS("port", "enable_tcp", "stream_id");
 
-        RtpServer::Ptr server = std::make_shared<RtpServer>();
-        server->start(allArgs["port"], allArgs["stream_id"], allArgs["enable_tcp"].as<bool>());
+        auto stream_id = allArgs["stream_id"];
 
-        auto port = server->getPort();
-        server->setOnDetach([port]() {
+        lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
+        if(s_rtpServerMap.find(stream_id) != s_rtpServerMap.end()) {
+            //为了防止RtpProcess所有权限混乱的问题，不允许重复添加相同的stream_id
+            throw InvalidArgsException("该stream_id已存在");
+        }
+
+        RtpServer::Ptr server = std::make_shared<RtpServer>();
+        server->start(allArgs["port"], stream_id, allArgs["enable_tcp"].as<bool>());
+        server->setOnDetach([stream_id]() {
             //设置rtp超时移除事件
             lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
-            s_rtpServerMap.erase(port);
+            s_rtpServerMap.erase(stream_id);
         });
 
         //保存对象
-        lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
-        s_rtpServerMap.emplace(port, server);
-
+        s_rtpServerMap.emplace(stream_id, server);
         //回复json
-        val["port"] = port;
+        val["port"] = server->getPort();
     });
 
     api_regist1("/index/api/closeRtpServer",[](API_ARGS1){
         CHECK_SECRET();
-        CHECK_ARGS("port");
+        CHECK_ARGS("stream_id");
 
         lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
-        val["hit"] = (int)s_rtpServerMap.erase(allArgs["port"].as<uint16_t>());
+        val["hit"] = (int) s_rtpServerMap.erase(allArgs["stream_id"]);
     });
 
     api_regist1("/index/api/listRtpServer",[](API_ARGS1){
         CHECK_SECRET();
 
         lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
-        for(auto &pr : s_rtpServerMap){
-            val["data"].append(pr.first);
+        for (auto &pr : s_rtpServerMap) {
+            Value obj;
+            obj["stream_id"] = pr.first;
+            obj["port"] = pr.second->getPort();
+            val["data"].append(obj);
         }
     });
 

src/Extension/Factory.cpp

@@ -118,7 +118,7 @@ RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
         case CodecH265 : return std::make_shared<H265RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
         case CodecAAC : return std::make_shared<AACRtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
         case CodecG711A :
-        case CodecG711U : return std::make_shared<G711RtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
+        case CodecG711U : return std::make_shared<G711RtpEncoder>(codec_id, ssrc, mtu, sample_rate, pt, interleaved);
         default : WarnL << "暂不支持该CodecId:" << codec_id; return nullptr;
     }
 }
@@ -129,7 +129,7 @@ RtpCodec::Ptr Factory::getRtpDecoderByTrack(const Track::Ptr &track) {
         case CodecH265 : return std::make_shared<H265RtpDecoder>();
         case CodecAAC : return std::make_shared<AACRtpDecoder>(track->clone());
         case CodecG711A :
-        case CodecG711U : return std::make_shared<G711RtpDecoder>(track->clone());
+        case CodecG711U : return std::make_shared<G711RtpDecoder>(track->getCodecId());
         default : WarnL << "暂不支持该CodecId:" << track->getCodecName(); return nullptr;
     }
 }

src/Extension/G711Rtp.cpp

@@ -12,8 +12,8 @@
 
 namespace mediakit{
 
-G711RtpDecoder::G711RtpDecoder(const Track::Ptr &track){
-    _codecid = track->getCodecId();
+G711RtpDecoder::G711RtpDecoder(CodecId codecid){
+    _codecid = codecid;
     _frame = obtainFrame();
 }
 
@@ -59,16 +59,10 @@ void G711RtpDecoder::onGetG711(const G711Frame::Ptr &frame) {
 
 /////////////////////////////////////////////////////////////////////////////////////
 
-G711RtpEncoder::G711RtpEncoder(uint32_t ui32Ssrc,
-                               uint32_t ui32MtuSize,
-                               uint32_t ui32SampleRate,
-                               uint8_t ui8PayloadType,
-                               uint8_t ui8Interleaved) :
-        RtpInfo(ui32Ssrc,
-                ui32MtuSize,
-                ui32SampleRate,
-                ui8PayloadType,
-                ui8Interleaved) {
+G711RtpEncoder::G711RtpEncoder(CodecId codecid, uint32_t ui32Ssrc, uint32_t ui32MtuSize,
+                               uint32_t ui32SampleRate, uint8_t ui8PayloadType, uint8_t ui8Interleaved) :
+                               G711RtpDecoder(codecid),
+                               RtpInfo(ui32Ssrc, ui32MtuSize, ui32SampleRate, ui8PayloadType, ui8Interleaved) {
 }
 
 void G711RtpEncoder::inputFrame(const Frame::Ptr &frame) {

src/Extension/G711Rtp.h

@@ -21,7 +21,7 @@ class G711RtpDecoder : public RtpCodec , public ResourcePoolHelper<G711Frame> {
 public:
     typedef std::shared_ptr<G711RtpDecoder> Ptr;
 
-    G711RtpDecoder(const Track::Ptr &track);
+    G711RtpDecoder(CodecId codecid);
     ~G711RtpDecoder() {}
 
     /**
@@ -35,9 +35,6 @@ public:
         return _codecid;
     }
 
-protected:
-    G711RtpDecoder() {}
-
 private:
     void onGetG711(const G711Frame::Ptr &frame);
     G711Frame::Ptr obtainFrame();
@@ -61,7 +58,8 @@ public:
      * @param ui8PayloadType pt类型
      * @param ui8Interleaved rtsp interleaved 值
      */
-    G711RtpEncoder(uint32_t ui32Ssrc,
+    G711RtpEncoder(CodecId codecid,
+                   uint32_t ui32Ssrc,
                    uint32_t ui32MtuSize,
                    uint32_t ui32SampleRate,
                    uint8_t ui8PayloadType = 0,

src/Rtp/RtpSelector.cpp

@@ -80,7 +80,9 @@ void RtpSelector::delProcess(const string &stream_id,const RtpProcess *ptr) {
     if (it->second->getProcess().get() != ptr) {
         return;
     }
+    auto process = it->second->getProcess();
     _map_rtp_process.erase(it);
+    process->onDetach();
 }
 
 void RtpSelector::onManager() {

src/Rtsp/RtspSession.h

@@ -59,51 +59,31 @@ public:
     //在请求明文密码时如果提供md5密码者则会导致认证失败
     typedef std::function<void(bool encrypted,const string &pwd_or_md5)> onAuth;
 
-    RtspSession(const Socket::Ptr &pSock);
+    RtspSession(const Socket::Ptr &sock);
     virtual ~RtspSession();
     ////TcpSession override////
-    void onRecv(const Buffer::Ptr &pBuf) override;
+    void onRecv(const Buffer::Ptr &buf) override;
     void onError(const SockException &err) override;
     void onManager() override;
+
 protected:
-    //RtspSplitter override
-    /**
-     * 收到完整的rtsp包回调，包括sdp等content数据
-     * @param parser rtsp包
-     */
+    /////RtspSplitter override/////
+    //收到完整的rtsp包回调，包括sdp等content数据
     void onWholeRtspPacket(Parser &parser) override;
-
-    /**
-     * 收到rtp包回调
-     * @param data
-     * @param len
-     */
-    void onRtpPacket(const char *data,uint64_t len) override;
-
-    /**
-     * 从rtsp头中获取Content长度
-     * @param parser
-     * @return
-     */
+    //收到rtp包回调
+    void onRtpPacket(const char *data, uint64_t len) override;
+    //从rtsp头中获取Content长度
     int64_t getContentLength(Parser &parser) override;
-
-    //RtpReceiver override
-    void onRtpSorted(const RtpPacket::Ptr &rtppt, int trackidx) override;
-    //MediaSourceEvent override
-    bool close(MediaSource &sender,bool force) override ;
+    ////RtpReceiver override////
+    void onRtpSorted(const RtpPacket::Ptr &rtp, int track_idx) override;
+    ////MediaSourceEvent override////
+    bool close(MediaSource &sender, bool force) override ;
     int totalReaderCount(MediaSource &sender) override;
-
-    //TcpSession override
+    /////TcpSession override////
     int send(const Buffer::Ptr &pkt) override;
+    //收到RTCP包回调
+    virtual void onRtcpPacket(int track_idx, SdpTrack::Ptr &track, unsigned char *data, unsigned int len);
 
-    /**
-     * 收到RTCP包回调
-     * @param iTrackidx
-     * @param track
-     * @param pucData
-     * @param uiLen
-     */
-    virtual void onRtcpPacket(int iTrackidx, SdpTrack::Ptr &track, unsigned char *pucData, unsigned int uiLen);
 private:
     //处理options方法,获取服务器能力
     void handleReq_Options(const Parser &parser);
@@ -127,101 +107,100 @@ private:
     void handleReq_Post(const Parser &parser);
     //处理SET_PARAMETER、GET_PARAMETER方法，一般用于心跳
     void handleReq_SET_PARAMETER(const Parser &parser);
-
     //rtsp资源未找到
-    void inline send_StreamNotFound();
+    void send_StreamNotFound();
     //不支持的传输模式
-    void inline send_UnsupportedTransport();
+    void send_UnsupportedTransport();
     //会话id错误
-    void inline send_SessionNotFound();
+    void send_SessionNotFound();
     //一般rtsp服务器打开端口失败时触发
-    void inline send_NotAcceptable();
-
+    void send_NotAcceptable();
     //获取track下标
-    inline int getTrackIndexByTrackType(TrackType type);
-    inline int getTrackIndexByControlSuffix(const string &controlSuffix);
-    inline int getTrackIndexByInterleaved(int interleaved);
-
+    int getTrackIndexByTrackType(TrackType type);
+    int getTrackIndexByControlSuffix(const string &control_suffix);
+    int getTrackIndexByInterleaved(int interleaved);
     //一般用于接收udp打洞包，也用于rtsp推流
-    inline void onRcvPeerUdpData(int intervaled, const Buffer::Ptr &pBuf, const struct sockaddr &addr);
+    void onRcvPeerUdpData(int interleaved, const Buffer::Ptr &buf, const struct sockaddr &addr);
     //配合onRcvPeerUdpData使用
-    inline void startListenPeerUdpData(int iTrackIdx);
-
+    void startListenPeerUdpData(int track_idx);
     ////rtsp专有认证相关////
     //认证成功
     void onAuthSuccess();
     //认证失败
-    void onAuthFailed(const string &realm,const string &why,bool close = true);
+    void onAuthFailed(const string &realm, const string &why, bool close = true);
     //开始走rtsp专有认证流程
-    void onAuthUser(const string &realm,const string &authorization);
+    void onAuthUser(const string &realm, const string &authorization);
     //校验base64方式的认证加密
-    void onAuthBasic(const string &realm,const string &strBase64);
+    void onAuthBasic(const string &realm, const string &auth_base64);
     //校验md5方式的认证加密
-    void onAuthDigest(const string &realm,const string &strMd5);
+    void onAuthDigest(const string &realm, const string &auth_md5);
     //触发url鉴权事件
     void emitOnPlay();
-
     //发送rtp给客户端
     void sendRtpPacket(const RtspMediaSource::RingDataType &pkt);
+    //触发rtcp发送
+    void onSendRtpPacket(const RtpPacket::Ptr &rtp);
     //回复客户端
-    bool sendRtspResponse(const string &res_code,const std::initializer_list<string> &header, const string &sdp = "" , const char *protocol = "RTSP/1.0");
-    bool sendRtspResponse(const string &res_code,const StrCaseMap &header = StrCaseMap(), const string &sdp = "",const char *protocol = "RTSP/1.0");
+    bool sendRtspResponse(const string &res_code, const std::initializer_list<string> &header, const string &sdp = "", const char *protocol = "RTSP/1.0");
+    bool sendRtspResponse(const string &res_code, const StrCaseMap &header = StrCaseMap(), const string &sdp = "", const char *protocol = "RTSP/1.0");
     //服务器发送rtcp
-    void sendSenderReport(bool overTcp,int iTrackIndex);
+    void sendSenderReport(bool over_tcp, int track_idx);
     //设置socket标志
     void setSocketFlags();
+
 private:
-    //用于判断客户端是否超时
-    Ticker _ticker;
+    //是否已经触发on_play事件
+    bool _emit_on_play = false;
+    //是否开始发送rtp
+    bool _enable_send_rtp;
+    //推流或拉流客户端采用的rtp传输方式
+    Rtsp::eRtpType _rtp_type = Rtsp::RTP_Invalid;
     //收到的seq，回复时一致
-    int _iCseq = 0;
+    int _cseq = 0;
+    //消耗的总流量
+    uint64_t _bytes_usage = 0;
     //ContentBase
-    string _strContentBase;
+    string _content_base;
     //Session号
-    string _strSession;
+    string _sessionid;
     //记录是否需要rtsp专属鉴权，防止重复触发事件
     string _rtsp_realm;
-    //是否已经触发on_play事件
-    bool _emit_on_play = false;
+    //登录认证
+    string _auth_nonce;
+    //用于判断客户端是否超时
+    Ticker _alive_ticker;
+
     //url解析后保存的相关信息
-    MediaInfo _mediaInfo;
+    MediaInfo _media_info;
+    //rtsp推流相关绑定的源
+    RtspMediaSourceImp::Ptr _push_src;
     //rtsp播放器绑定的直播源
-    std::weak_ptr<RtspMediaSource> _pMediaSrc;
+    std::weak_ptr<RtspMediaSource> _play_src;
     //直播源读取器
-    RtspMediaSource::RingType::RingReader::Ptr _pRtpReader;
-    //推流或拉流客户端采用的rtp传输方式
-    Rtsp::eRtpType _rtpType = Rtsp::RTP_Invalid;
+    RtspMediaSource::RingType::RingReader::Ptr _play_reader;
     //sdp里面有效的track,包含音频或视频
-    vector<SdpTrack::Ptr> _aTrackInfo;
+    vector<SdpTrack::Ptr> _sdp_track;
+
+    //rtcp统计,trackid idx 为数组下标
+    RtcpCounter _rtcp_counter[2];
+    //rtcp发送时间,trackid idx 为数组下标
+    Ticker _rtcp_send_tickers[2];
+
     ////////RTP over udp////////
     //RTP端口,trackid idx 为数组下标
-    Socket::Ptr _apRtpSock[2];
+    Socket::Ptr _rtp_socks[2];
     //RTCP端口,trackid idx 为数组下标
-    Socket::Ptr _apRtcpSock[2];
+    Socket::Ptr _rtcp_socks[2];
     //标记是否收到播放的udp打洞包,收到播放的udp打洞包后才能知道其外网udp端口号
-    unordered_set<int> _udpSockConnected;
+    unordered_set<int> _udp_connected_flags;
     ////////RTP over udp_multicast////////
     //共享的rtp组播对象
     RtpMultiCaster::Ptr _multicaster;
-
-    //登录认证
-    string _strNonce;
-    //消耗的总流量
-    uint64_t _ui64TotalBytes = 0;
-
-    //RTSP over HTTP
+    ////////RTSP over HTTP  ////////
     //quicktime 请求rtsp会产生两次tcp连接，
     //一次发送 get 一次发送post，需要通过x-sessioncookie关联起来
     string _http_x_sessioncookie;
-    function<void(const Buffer::Ptr &pBuf)> _onRecv;
-    //是否开始发送rtp
-    bool _enableSendRtp;
-    //rtsp推流相关
-    RtspMediaSourceImp::Ptr _pushSrc;
-    //rtcp统计,trackid idx 为数组下标
-    RtcpCounter _aRtcpCnt[2];
-    //rtcp发送时间,trackid idx 为数组下标
-    Ticker _aRtcpTicker[2];
+    function<void(const Buffer::Ptr &)> _on_recv;
 };
 
 /**