优化rtsp客户端对点播的支持 (#4575 #4569 #4576)

Co-authored-by: xia-chu <771730766@qq.com>

3rdpart/ZLToolKit

@@ -1 +1 @@
-Subproject commit 84a853154215d045be2edb569a357c263e941bf1
+Subproject commit 493d14e1682cc0c79e72b3893dd98f865d45b8e9

conf/config.ini

@@ -338,7 +338,7 @@ gop_cache=1
 
 #国标发送g711 rtp 打包时，每个包的语音时长是多少，默认是100 ms，范围为20~180ms (gb28181-2016，c.2.4规定)，
 #最好为20 的倍数，程序自动向20的倍数取整
-rtp_g711_dur_ms = 100
+rtp_g711_dur_ms=100
 #udp接收数据socket buffer大小配置
 #4*1024*1024=4196304
 udp_recv_socket_buffer=4194304
@@ -359,8 +359,12 @@ iceTransportPolicy=0
 #STUN/TURN 服务Ice密码
 iceUfrag=ZLMediaKit
 icePwd=ZLMediaKit
+#webrtc datachannel是否回显数据，测试用
+datachannel_echo=1
+
+max_stun_retry=7
 #TURN服务分配端口池
-portRange=50000-65000
+port_range=49152-65535
 #rtc播放推流、播放超时时间
 timeoutSec=15
 #本机对rtc客户端的可见ip，作为服务器时一般为公网ip，可有多个，用','分开，当置空时，会自动获取网卡ip
@@ -426,7 +430,6 @@ pktBufSize=8192
 #srt udp服务器的密码,为空表示不加密
 passPhrase=
 
-
 [rtsp]
 #rtsp专有鉴权方式是采用base64还是md5方式
 authBasic=0
@@ -453,9 +456,13 @@ lowLatency=0
 #当客户端发起RTSP SETUP的时候如果传输类型和此配置不一致则返回461 Unsupported transport
 #迫使客户端重新SETUP并切换到对应协议。目前支持FFMPEG和VLC
 rtpTransportType=-1
+
 [shell]
 #调试telnet服务器接受最大buffer大小
 maxReqSize=1024
 #调试telnet服务器监听端口
 port=0
 
+# onvif搜索用
+[onvif]
+port=3702

src/Common/config.cpp

@@ -30,7 +30,22 @@ bool loadIniConfig(const char *ini_path) {
         ini = exePath() + ".ini";
     }
     try {
-        mINI::Instance().parseFile(ini);
+        mINI tmp;
+        tmp.parseFile(ini);
+
+        auto &ref = mINI::Instance();
+        for (auto &pr : tmp) {
+            if (ref.find(pr.first) == ref.end()) {
+                // 新增键
+                WarnL << "unknow config: " << pr.first << " = " << pr.second;
+                ref.emplace(pr);
+            } else {
+                // 更新键
+                ref[pr.first] = pr.second;
+            }
+        }
+        // 更新注释和排序
+        ref.updateFrom(tmp);
         NOTICE_EMIT(BroadcastReloadConfigArgs, Broadcast::kBroadcastReloadConfig);
         return true;
     } catch (std::exception &) {

src/Rtsp/RtspPlayer.cpp

@@ -464,7 +464,7 @@ void RtspPlayer::sendPause(int type, uint32_t seekMS) {
     // Start or pause RTSP
     switch (type) {
         case type_pause: sendRtspRequest("PAUSE", _control_url, {}); break;
-        case type_play: sendRtspRequest("PLAY", _content_base); break;
+        case type_play:
         case type_seek:
             sendRtspRequest("PLAY", _control_url, { "Range", StrPrinter << "npt=" << setiosflags(ios::fixed) << setprecision(2) << seekMS / 1000.0 << "-" });
             break;
@@ -582,6 +582,9 @@ void RtspPlayer::onRtcpPacket(int track_idx, SdpTrack::Ptr &track, uint8_t *data
 
 void RtspPlayer::onRtpSorted(RtpPacket::Ptr rtppt, int trackidx) {
     _stamp[trackidx] = rtppt->getStampMS();
+    if (!_first_stamp[trackidx]) {
+        _first_stamp[trackidx] = _stamp[trackidx];
+    }
     _rtp_recv_ticker.resetTime();
     onRecvRTP(std::move(rtppt), _sdp_track[trackidx]);
 }
@@ -609,7 +612,7 @@ float RtspPlayer::getPacketLossRate(TrackType type) const {
 }
 
 uint32_t RtspPlayer::getProgressMilliSecond() const {
-    return MAX(_stamp[0], _stamp[1]);
+    return MAX(_stamp[0] - _first_stamp[0], _stamp[1] - _first_stamp[1]);
 }
 
 void RtspPlayer::seekToMilliSecond(uint32_t ms) {
@@ -685,14 +688,18 @@ void RtspPlayer::sendRtspRequest(const string &cmd, const string &url, const Str
     printer << cmd << " " << url << " RTSP/1.0\r\n";
 
     TraceL << cmd << " "<< url;
+
+    if (cmd == "PLAY") {
+        // play命令时支持覆盖更新rtsp头，用于onvif点播等场景
+        for (auto &pr : _custom_header) {
+            header[pr.first] = pr.second;
+        }
+    }
+
     for (auto &pr : header) {
         printer << pr.first << ": " << pr.second << "\r\n";
     }
-    if (cmd == "PLAY") {
-        for (auto &pr : _custom_header) {
-            printer << pr.first << ": " << pr.second << "\r\n";
-        }
-    }
+
     printer << "\r\n";
     SockSender::send(std::move(printer));
 }

src/Rtsp/RtspPlayer.h

@@ -184,9 +184,12 @@ protected:
     Rtsp::eRtpType _rtp_type = Rtsp::RTP_TCP;
 
 private:
+    // 起始时间戳
+    uint64_t _first_stamp[2] = {0, 0};
+
     // 当前rtp时间戳  [AUTO-TRANSLATED:410f2691]
     // Current rtp timestamp
-    uint32_t _stamp[2] = {0, 0};
+    uint64_t _stamp[2] = {0, 0};
 
     // 超时功能实现  [AUTO-TRANSLATED:1d603b3a]
     // Timeout function implementation