优化rtsp服务器代码

修复一个rtp over http的bug

优化rtsp服务器代码
修复一个rtp over http的bug
f411ddc2 · xiongziliang · 91d88887 · f411ddc2 · f411ddc2 · f411ddc2
Commit f411ddc2 authored Dec 14, 2018 by xiongziliang
--- a/src/Common/config.h
+++ b/src/Common/config.h
@@ -57,14 +57,12 @@ bool loadIniConfig(const char *ini_path = nullptr);
 #define CLEAR_ARR(arr) for(auto &item : arr){ item = 0;}
 #endif //CLEAR_ARR
-#define SERVER_NAME "ZLMediaKit"
+#define SERVER_NAME "ZLMediaKit-3.0"
 #define VHOST_KEY "vhost"
 #define HTTP_SCHEMA "http"
 #define RTSP_SCHEMA "rtsp"
 #define RTMP_SCHEMA "rtmp"
 #define DEFAULT_VHOST "__defaultVhost__"
-#define RTSP_VERSION 1.30
-#define RTSP_BUILDTIME __DATE__" CST"
 ////////////广播名称///////////
 namespace Broadcast {

--- a/src/Http/HttpRequestSplitter.cpp
+++ b/src/Http/HttpRequestSplitter.cpp
@@ -54,12 +54,16 @@ void HttpRequestSplitter::input(const char *data,uint64_t len) {
    //数据按照请求头处理
    const char *index = nullptr;
-    uint64_t remain = len;
+    _remain_data_size = len;
-    while (_content_len == 0 && remain > 0 && (index = onSearchPacketTail(ptr,remain)) != nullptr) {
+    while (_content_len == 0 && _remain_data_size > 0 && (index = onSearchPacketTail(ptr,_remain_data_size)) != nullptr) {
        //_content_len == 0，这是请求头
-        _content_len = onRecvHeader(ptr, index - ptr);
+        const char *header_ptr = ptr;
+        int64_t header_size = index - ptr;
        ptr = index;
-        remain = len - (ptr - data);
+        _remain_data_size = len - (ptr - data);
+        _content_len = onRecvHeader(header_ptr, header_size);
    }
    /*
@@ -67,7 +71,7 @@ void HttpRequestSplitter::input(const char *data,uint64_t len) {
     */
    tail_ref = tail_tmp;
-    if(remain <= 0){
+    if(_remain_data_size <= 0){
        //没有剩余数据，清空缓存
        _remain_data.clear();
        return;
@@ -75,7 +79,7 @@ void HttpRequestSplitter::input(const char *data,uint64_t len) {
    if(_content_len == 0){
        //尚未找到http头，缓存定位到剩余数据部分
-        string str(ptr,remain);
+        string str(ptr,_remain_data_size);
        _remain_data = str;
        return;
    }
@@ -83,23 +87,23 @@ void HttpRequestSplitter::input(const char *data,uint64_t len) {
    //已经找到http头了
    if(_content_len > 0){
        //数据按照固定长度content处理
-        if(remain < _content_len){
+        if(_remain_data_size < _content_len){
            //数据不够，缓存定位到剩余数据部分
-            string str(ptr,remain);
+            string str(ptr,_remain_data_size);
            _remain_data = str;
            return;
        }
        //收到content数据，并且接受content完毕
        onRecvContent(ptr,_content_len);
-        remain -= _content_len;
+        _remain_data_size -= _content_len;
        ptr += _content_len;
        //content处理完毕,后面数据当做请求头处理
        _content_len = 0;
-        if(remain > 0){
+        if(_remain_data_size > 0){
            //还有数据没有处理完毕
-            string str(ptr,remain);
+            string str(ptr,_remain_data_size);
            _remain_data = str;
            data = ptr = (char *)_remain_data.data();
@@ -112,7 +116,7 @@ void HttpRequestSplitter::input(const char *data,uint64_t len) {
    //_content_len < 0;数据按照不固定长度content处理
-    onRecvContent(ptr,remain);//消费掉所有剩余数据
+    onRecvContent(ptr,_remain_data_size);//消费掉所有剩余数据
    _remain_data.clear();
 }
@@ -133,6 +137,10 @@ const char *HttpRequestSplitter::onSearchPacketTail(const char *data,int len) {
    return  pos + 4;
 }
+int64_t HttpRequestSplitter::remainDataSize() {
+    return _remain_data_size;
+}
 } /* namespace mediakit */
--- a/src/Http/HttpRequestSplitter.h
+++ b/src/Http/HttpRequestSplitter.h
@@ -81,9 +81,15 @@ protected:
     * 恢复初始设置
     */
     void reset();
+     /**
+      * 剩余数据大小
+      */
+     int64_t remainDataSize();
 private:
    string _remain_data;
    int64_t _content_len = 0;
+    int64_t _remain_data_size = 0;
 };
 } /* namespace mediakit */

--- a/src/Player/PlayerBase.h
+++ b/src/Player/PlayerBase.h
@@ -87,6 +87,7 @@ class PlayerBase : public DemuxerBase, public mINI{
 public:
 	typedef std::shared_ptr<PlayerBase> Ptr;
 	typedef enum {
+		RTP_Invalid = -1,
 		RTP_TCP = 0,
 		RTP_UDP = 1,
 		RTP_MULTICAST = 2,

--- a/src/Rtsp/RtspSession.cpp
+++ b/src/Rtsp/RtspSession.cpp
--- a/src/Rtsp/RtspSession.h
+++ b/src/Rtsp/RtspSession.h
@@ -77,27 +77,24 @@ public:
 	void onRecv(const Buffer::Ptr &pBuf) override;
 	void onError(const SockException &err) override;
 	void onManager() override;
 protected:
 	//HttpRequestSplitter override
 	int64_t onRecvHeader(const char *data,uint64_t len) override ;
+	void onRecvContent(const char *data,uint64_t len) override;
 private:
 	void inputRtspOrRtcp(const char *data,uint64_t len);
-	int send(const Buffer::Ptr &pkt) override{
-        _ui64TotalBytes += pkt->size();
-        return _pSender->send(pkt,_flags);
-	}
 	void shutdown() override ;
 	void shutdown_l(bool close);
-	bool handleReq_Options(); //处理options方法
+	int handleReq_Options(); //处理options方法
-	bool handleReq_Describe(); //处理describe方法
+	int handleReq_Describe(); //处理describe方法
-	bool handleReq_Setup(); //处理setup方法
+	int handleReq_ANNOUNCE(); //处理options方法
-	bool handleReq_Play(); //处理play方法
+	int handleReq_Setup(); //处理setup方法
-	bool handleReq_Pause(); //处理pause方法
+	int handleReq_Play(); //处理play方法
-	bool handleReq_Teardown(); //处理teardown方法
+	int handleReq_Pause(); //处理pause方法
-	bool handleReq_Get(); //处理Get方法
+	int handleReq_Teardown(); //处理teardown方法
-	bool handleReq_Post(); //处理Post方法
+	int handleReq_Get(); //处理Get方法
-	bool handleReq_SET_PARAMETER(); //处理SET_PARAMETER方法
+	int handleReq_Post(); //处理Post方法
+	int handleReq_SET_PARAMETER(); //处理SET_PARAMETER方法
 	void inline send_StreamNotFound(); //rtsp资源未找到
 	void inline send_UnsupportedTransport(); //不支持的传输模式
@@ -106,33 +103,9 @@ private:
 	inline bool findStream(); //根据rtsp url查找 MediaSource实例
    inline void findStream(const function<void(bool)> &cb); //根据rtsp url查找 MediaSource实例
-	inline void initSender(const std::shared_ptr<RtspSession> &pSession); //处理rtsp over http，quicktime使用的
+	inline string printSSRC(uint32_t ui32Ssrc);
-	inline void sendRtpPacket(const RtpPacket::Ptr &pkt);
+	inline int getTrackIndexByTrackType(TrackType type);
-	inline string printSSRC(uint32_t ui32Ssrc) {
+    inline int getTrackIndexByControlSuffix(const string &controlSuffix);
-		char tmp[9] = { 0 };
-		ui32Ssrc = htonl(ui32Ssrc);
-		uint8_t *pSsrc = (uint8_t *) &ui32Ssrc;
-		for (int i = 0; i < 4; i++) {
-			sprintf(tmp + 2 * i, "%02X", pSsrc[i]);
-		}
-		return tmp;
-	}
-	inline int getTrackIndexByTrackType(TrackType type) {
-		for (unsigned int i = 0; i < _aTrackInfo.size(); i++) {
-			if (type == _aTrackInfo[i]->_type) {
-				return i;
-			}
-		}
-		return -1;
-	}
-    inline int getTrackIndexByControlSuffix(const string &controlSuffix) {
-        for (unsigned int i = 0; i < _aTrackInfo.size(); i++) {
-            if (controlSuffix == _aTrackInfo[i]->_control_surffix) {
-                return i;
-            }
-        }
-        return -1;
-    }
 	inline void onRcvPeerUdpData(int iTrackIdx, const Buffer::Ptr &pBuf, const struct sockaddr &addr);
 	inline void startListenPeerUdpData();
@@ -146,10 +119,16 @@ private:
    void doDelay(int delaySec,const std::function<void()> &fun);
    void cancelDelyaTask();
+	inline void sendRtpPacket(const RtpPacket::Ptr &pkt);
+	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");
+	int send(const Buffer::Ptr &pkt) override;
+	inline void initSender(const std::shared_ptr<RtspSession> &pSession); //处理rtsp over http，quicktime使用的
 private:
-	char *_pcBuf = nullptr;
 	Ticker _ticker;
 	Parser _parser; //rtsp解析类
+	int _iCseq = 0;
 	string _strUrl;
 	string _strSdp;
 	string _strSession;
@@ -157,31 +136,23 @@ private:
    MediaInfo _mediaInfo;
 	std::weak_ptr<RtspMediaSource> _pMediaSrc;
 	RingBuffer<RtpPacket::Ptr>::RingReader::Ptr _pRtpReader;
+	PlayerBase::eRtpType _rtpType = PlayerBase::RTP_Invalid;
-	PlayerBase::eRtpType _rtpType = PlayerBase::RTP_UDP;
-	bool _bSetUped = false;
-	int _iCseq = 0;
-	SdpAttr _sdpAttr;
 	vector<SdpTrack::Ptr> _aTrackInfo;
+	//RTP over udp
 	bool _bGotAllPeerUdp = false;
-#ifdef RTSP_SEND_RTCP
-	RtcpCounter _aRtcpCnt[2]; //rtcp统计,trackid idx 为数组下标
-	Ticker _aRtcpTicker[2]; //rtcp发送时间,trackid idx 为数组下标
-	inline void sendRTCP();
-#endif
-	//RTP over UDP
 	bool _abGotPeerUdp[2] = { false, false }; //获取客户端udp端口计数
 	weak_ptr<Socket> _apUdpSock[2]; //发送RTP的UDP端口,trackid idx 为数组下标
 	std::shared_ptr<struct sockaddr> _apPeerUdpAddr[2]; //播放器接收RTP的地址,trackid idx 为数组下标
 	bool _bListenPeerUdpData = false;
+	//RTP over udp_multicast
 	RtpBroadCaster::Ptr _pBrdcaster;
 	//登录认证
    string _strNonce;
+    //消耗的总流量
+    uint64_t _ui64TotalBytes = 0;
 	//RTSP over HTTP
 	function<void(void)> _onDestory;
@@ -190,18 +161,21 @@ private:
 	//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sessioncookie关联起来
 	string _strSessionCookie;
-    //消耗的总流量
-    uint64_t _ui64TotalBytes = 0;
 	static recursive_mutex g_mtxGetter; //对quicktime上锁保护
 	static recursive_mutex g_mtxPostter; //对quicktime上锁保护
 	static unordered_map<string, weak_ptr<RtspSession> > g_mapGetter;
 	static unordered_map<void *, std::shared_ptr<RtspSession> > g_mapPostter;
+	function<void(const char *data,uint64_t len)> _onContent;
    std::function<void()> _delayTask;
    uint32_t _iTaskTimeLine = 0;
    atomic<bool> _enableSendRtp;
+#ifdef RTSP_SEND_RTCP
+	RtcpCounter _aRtcpCnt[2]; //rtcp统计,trackid idx 为数组下标
+	Ticker _aRtcpTicker[2]; //rtcp发送时间,trackid idx 为数组下标
+	inline void sendRTCP();
+#endif
 };
 } /* namespace mediakit */