Merge branch 'master' into dev

.github/workflows/docker.yml

@@ -40,9 +40,7 @@ jobs:
       # https://github.com/sigstore/cosign-installer
       - name: Install cosign
         if: github.event_name != 'pull_request'
-        uses: sigstore/cosign-installer@d6a3abf1bdea83574e28d40543793018b6035605
+        uses: sigstore/cosign-installer@d572c9c13673d2e0a26fabf90b5748f36886883f
-        with:
-          cosign-release: 'v1.7.1'
 
       - name: Set up QEMU
         uses: docker/setup-qemu-action@v2

.gitmodules

@@ -7,3 +7,6 @@
 [submodule "3rdpart/jsoncpp"]
 	path = 3rdpart/jsoncpp
 	url = https://gitee.com/mirrors/jsoncpp.git
+[submodule "www/webassist"]
+	path = www/webassist
+	url = https://gitee.com/victor1002/zlm_webassist

.gitmodules_github

@@ -7,3 +7,6 @@
 [submodule "3rdpart/jsoncpp"]
     path = 3rdpart/jsoncpp
     url = https://github.com/open-source-parsers/jsoncpp.git
+[submodule "www/webassist"]
+	path = www/webassist
+	url = https://github.com/1002victor/zlm_webassist

3rdpart/ZLToolKit

@@ -1 +1 @@
-Subproject commit b11582c38e8dbbb8d93ca9ce33c9a0b0cd58f59a
+Subproject commit 97871cfa78fcd2fae164243a8c653e323385772d

3rdpart/jsoncpp

@@ -1 +1 @@
-Subproject commit 8190e061bc2d95da37479a638aa2c9e483e58ec6
+Subproject commit 69098a18b9af0c47549d9a271c054d13ca92b006

3rdpart/media-server

@@ -1 +1 @@
-Subproject commit cdbb3d6b9ea254f454c6e466c5962af5ace01199
+Subproject commit 3dc623a899eee3810587fb267dbff770b626a55b

AUTHORS

@@ -85,3 +85,6 @@ WuPeng <wp@zafu.edu.cn>
 [linxiaoyan87](https://github.com/linxiaoyan)
 [waken](https://github.com/mc373906408)
 [Deepslient](https://github.com/Deepslient)
+[imp_rayjay](https://github.com/rayjay214)
+[ArmstrongCN](https://github.com/ArmstrongCN)
+[leibnewton](https://github.com/leibnewton)

README.md

@@ -184,6 +184,7 @@ bash build_docker_images.sh
    - [BXC_gb28181Player](https://github.com/any12345com/BXC_gb28181Player) C++开发的支持国标GB28181协议的视频流播放器
 
 - WEB管理网站
+   - [zlm_webassist](https://github.com/1002victor/zlm_webassist) 本项目配套的前后端分离web管理项目
    - [AKStreamNVR](https://github.com/langmansh/AKStreamNVR) 前后端分离web项目,支持webrtc播放
    
  - SDK
@@ -329,6 +330,10 @@ bash build_docker_images.sh
 [linxiaoyan87](https://github.com/linxiaoyan)
 [waken](https://github.com/mc373906408)
 [Deepslient](https://github.com/Deepslient)
+[imp_rayjay](https://github.com/rayjay214)
+[ArmstrongCN](https://github.com/ArmstrongCN)
+[leibnewton](https://github.com/leibnewton)
+[1002victor](https://github.com/1002victor)
 
 同时感谢JetBrains对开源项目的支持，本项目使用CLion开发与调试：
 

README_en.md

@@ -342,6 +342,7 @@ bash build_docker_images.sh
 ## Collaborative Projects
 
 - Visual management website
+   - [A backend management website for this project](https://github.com/1002victor/zlm_webassist)
    - [The latest web project with front-end and back-end separation, supporting webrtc playback](https://github.com/langmansh/AKStreamNVR)
    - [Management web site based on ZLMediaKit master branch](https://gitee.com/kkkkk5G/MediaServerUI) 
    - [Management web site based on ZLMediaKit branch](https://github.com/chenxiaolei/ZLMediaKit_NVR_UI)
@@ -493,6 +494,10 @@ Thanks to all those who have supported this project in various ways, including b
 [linxiaoyan87](https://github.com/linxiaoyan)
 [waken](https://github.com/mc373906408)
 [Deepslient](https://github.com/Deepslient)
+[imp_rayjay](https://github.com/rayjay214)
+[ArmstrongCN](https://github.com/ArmstrongCN)
+[leibnewton](https://github.com/leibnewton)
+[1002victor](https://github.com/1002victor)
 
 Also thank to JetBrains for their support for open source project, we developed and debugged zlmediakit with CLion:
 

postman/ZLMediaKit.postman_collection.json

@@ -1,9 +1,10 @@
 {
 	"info": {
-		"_postman_id": "39e8a1df-cc8e-4e3f-bf5e-197c86e7bf0f",
+		"_postman_id": "509e5f6b-728c-4d5f-b3e8-521d76b2cc7a",
 		"name": "ZLMediaKit",
 		"description": "媒体服务器",
-		"schema": "https://schema.getpostman.com/json/collection/v2.1.0/collection.json"
+		"schema": "https://schema.getpostman.com/json/collection/v2.1.0/collection.json",
+		"_exporter_id": "29185956"
 	},
 	"item": [
 		{
@@ -918,7 +919,7 @@
 				"method": "GET",
 				"header": [],
 				"url": {
-					"raw": "{{ZLMediaKit_URL}}/index/api/broadcastMessage?secret={{ZLMediaKit_secret}}&schema=rtsp&vhost={{defaultVhost}}&app=live&stream=test&msg=Hello zlmediakit123",
+					"raw": "{{ZLMediaKit_URL}}/index/api/broadcastMessage?secret={{ZLMediaKit_secret}}&schema=rtsp&vhost={{defaultVhost}}&app=live&stream=test&msg=Hello ZLMediakit",
 					"host": [
 						"{{ZLMediaKit_URL}}"
 					],
@@ -1247,7 +1248,7 @@
 						},
 						{
 							"key": "stamp",
-							"value": 1000,
+							"value": "1000",
 							"description": "要设置的录像播放位置"
 						}
 					]
@@ -1478,6 +1479,53 @@
 			},
 			"response": []
 		},
+		{
+			"name": "创建多路复用RTP服务器(openRtpServerMultiplex)",
+			"request": {
+				"method": "GET",
+				"header": [],
+				"url": {
+					"raw": "{{ZLMediaKit_URL}}/index/api/openRtpServer?secret={{ZLMediaKit_secret}}&port=0&tcp_mode=1&stream_id=test",
+					"host": [
+						"{{ZLMediaKit_URL}}"
+					],
+					"path": [
+						"index",
+						"api",
+						"openRtpServer"
+					],
+					"query": [
+						{
+							"key": "secret",
+							"value": "{{ZLMediaKit_secret}}",
+							"description": "api操作密钥(配置文件配置)"
+						},
+						{
+							"key": "port",
+							"value": "0",
+							"description": "绑定的端口，0时为随机端口"
+						},
+						{
+							"key": "tcp_mode",
+							"value": "1",
+							"description": "tcp模式，0时为不启用tcp监听，1时为启用tcp监听"
+						},
+						{
+							"key": "stream_id",
+							"value": "test",
+							"description": "该端口绑定的流id\n"
+						},
+						{
+							"key": "only_audio",
+							"value": "0",
+							"description": "是否为单音频track，用于语音对讲",
+							"disabled": true
+						}
+					]
+				}
+			},
+			"response": []
+		},
 		{
 			"name": "连接RTP服务器(connectRtpServer)",
 			"request": {
@@ -1710,10 +1758,16 @@
 							"value": "obs",
 							"description": "流id，例如 obs"
 						},
+						{
+							"key": "ssrc_multi_send",
+							"value": "0",
+							"description": "是否支持同ssrc推流到多个上级服务器,该参数非必选参数 默认false",
+							"disabled": true
+						},
 						{
 							"key": "ssrc",
 							"value": "1",
-							"description": "rtp推流的ssrc，ssrc不同时，可以推流到多个上级服务器"
+							"description": "rtp推流的ssrc"
 						},
 						{
 							"key": "dst_url",

server/WebApi.cpp

@@ -370,6 +370,7 @@ Value makeMediaSourceJson(MediaSource &media){
             obj["loss"] = loss;
         }
         obj["frames"] = track->getFrames();
+        obj["duration"] = track->getDuration();
         switch(codec_type){
             case TrackAudio : {
                 auto audio_track = dynamic_pointer_cast<AudioTrack>(track);
@@ -403,7 +404,7 @@ Value makeMediaSourceJson(MediaSource &media){
 }
 
 #if defined(ENABLE_RTPPROXY)
-uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio) {
+uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mode, const string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex) {
     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
@@ -411,7 +412,7 @@ uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mod
     }
 
     RtpServer::Ptr server = std::make_shared<RtpServer>();
-    server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_audio);
+    server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_audio, multiplex);
     server->setOnDetach([stream_id]() {
         //设置rtp超时移除事件
         lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
@@ -1182,6 +1183,25 @@ void installWebApi() {
         val["port"] = port;
     });
 
+      api_regist("/index/api/openRtpServerMultiplex", [](API_ARGS_MAP) {
+      CHECK_SECRET();
+      CHECK_ARGS("port", "stream_id");
+      auto stream_id = allArgs["stream_id"];
+      auto tcp_mode = allArgs["tcp_mode"].as<int>();
+      if (allArgs["enable_tcp"].as<int>() && !tcp_mode) {
+          // 兼容老版本请求，新版本去除enable_tcp参数并新增tcp_mode参数
+          tcp_mode = 1;
+      }
+
+      auto port = openRtpServer(
+          allArgs["port"], stream_id, tcp_mode, "::", true, 0, allArgs["only_audio"].as<bool>(),true);
+      if (port == 0) {
+          throw InvalidArgsException("该stream_id已存在");
+      }
+      // 回复json
+      val["port"] = port;
+  });
+
     api_regist("/index/api/connectRtpServer", [](API_ARGS_MAP_ASYNC) {
         CHECK_SECRET();
         CHECK_ARGS("stream_id", "dst_url", "dst_port");
@@ -1244,6 +1264,7 @@ void installWebApi() {
         args.passive = false;
         args.dst_url = allArgs["dst_url"];
         args.dst_port = allArgs["dst_port"];
+        args.ssrc_multi_send = allArgs["ssrc_multi_send"].empty() ? false : allArgs["ssrc_multi_send"].as<bool>();
         args.ssrc = allArgs["ssrc"];
         args.is_udp = allArgs["is_udp"];
         args.src_port = allArgs["src_port"];
@@ -1491,11 +1512,15 @@ void installWebApi() {
     // http://127.0.0.1/index/api/deleteRecordDirectroy?vhost=__defaultVhost__&app=live&stream=ss&period=2020-01-01
     api_regist("/index/api/deleteRecordDirectory", [](API_ARGS_MAP) {
         CHECK_SECRET();
-        CHECK_ARGS("vhost", "app", "stream");
+        CHECK_ARGS("vhost", "app", "stream", "period");
         auto tuple = MediaTuple{allArgs["vhost"], allArgs["app"], allArgs["stream"]};
         auto record_path = Recorder::getRecordPath(Recorder::type_mp4, tuple, allArgs["customized_path"]);
         auto period = allArgs["period"];
         record_path = record_path + period + "/";
+        auto name = allArgs["name"];
+        if (!name.empty()) {
+            record_path += name;
+        }
         int result = File::delete_file(record_path.data());
         if (result) {
             // 不等于0时代表失败
@@ -1911,24 +1936,28 @@ void installWebApi() {
 void unInstallWebApi(){
     {
         lock_guard<recursive_mutex> lck(s_proxyMapMtx);
-        s_proxyMap.clear();
+        auto proxyMap(std::move(s_proxyMap));
+        proxyMap.clear();
     }
 
     {
         lock_guard<recursive_mutex> lck(s_ffmpegMapMtx);
-        s_ffmpegMap.clear();
+        auto ffmpegMap(std::move(s_ffmpegMap));
+        ffmpegMap.clear();
     }
 
     {
         lock_guard<recursive_mutex> lck(s_proxyPusherMapMtx);
-        s_proxyPusherMap.clear();
+        auto proxyPusherMap(std::move(s_proxyPusherMap));
+        proxyPusherMap.clear();
     }
 
     {
 #if defined(ENABLE_RTPPROXY)
         RtpSelector::Instance().clear();
         lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
-        s_rtpServerMap.clear();
+        auto rtpServerMap(std::move(s_rtpServerMap));
+        rtpServerMap.clear();
 #endif
     }
     NoticeCenter::Instance().delListener(&web_api_tag);

server/WebApi.h

@@ -239,7 +239,7 @@ void installWebApi();
 void unInstallWebApi();
 
 #if defined(ENABLE_RTPPROXY)
-uint16_t openRtpServer(uint16_t local_port, const std::string &stream_id, int tcp_mode, const std::string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio);
+uint16_t openRtpServer(uint16_t local_port, const std::string &stream_id, int tcp_mode, const std::string &local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex=false);
 void connectRtpServer(const std::string &stream_id, const std::string &dst_url, uint16_t dst_port, const std::function<void(const toolkit::SockException &ex)> &cb);
 bool closeRtpServer(const std::string &stream_id);
 #endif

server/WebHook.cpp

@@ -674,7 +674,7 @@ void installWebHook() {
         });
     });
 
-    NoticeCenter::Instance().addListener(&web_hook_tag, Broadcast::KBroadcastRtpServerTimeout, [](BroadcastRtpServerTimeoutArgs) {
+    NoticeCenter::Instance().addListener(&web_hook_tag, Broadcast::kBroadcastRtpServerTimeout, [](BroadcastRtpServerTimeoutArgs) {
         GET_CONFIG(string, rtp_server_timeout, Hook::kOnRtpServerTimeout);
         if (!hook_enable || rtp_server_timeout.empty()) {
             return;

src/Common/JemallocUtil.cpp

@@ -12,6 +12,7 @@
 #include "JemallocUtil.h"
 #include "Util/logger.h"
 #ifdef USE_JEMALLOC
+#include <array>
 #include <iostream>
 #include <jemalloc/jemalloc.h>
 #endif

src/Common/MediaSource.h

@@ -99,11 +99,13 @@ public:
         // rtp采用ps还是es方式
         bool use_ps = true;
         //发送es流时指定是否只发送纯音频流
-        bool only_audio = true;
+        bool only_audio = false;
         //tcp被动方式
         bool passive = false;
         // rtp payload type
         uint8_t pt = 96;
+        //是否支持同ssrc多服务器发送
+        bool ssrc_multi_send = false;
         // 指定rtp ssrc
         std::string ssrc;
         // 指定本地发送端口

src/Common/MultiMediaSourceMuxer.cpp

@@ -45,6 +45,7 @@ static string getTrackInfoStr(const TrackSource *track_src){
     _StrPrinter codec_info;
     auto tracks = track_src->getTracks(true);
     for (auto &track : tracks) {
+        track->update();
         auto codec_type = track->getTrackType();
         codec_info << track->getCodecName();
         switch (codec_type) {
@@ -294,12 +295,14 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &sender, const MediaSourceE
 
     auto ring = _ring;
     auto ssrc = args.ssrc;
+    auto ssrc_multi_send = args.ssrc_multi_send;
     auto tracks = getTracks(false);
     auto poller = getOwnerPoller(sender);
     auto rtp_sender = std::make_shared<RtpSender>(poller);
+
     weak_ptr<MultiMediaSourceMuxer> weak_self = shared_from_this();
 
-    rtp_sender->startSend(args, [ssrc, weak_self, rtp_sender, cb, tracks, ring, poller](uint16_t local_port, const SockException &ex) mutable {
+    rtp_sender->startSend(args, [ssrc,ssrc_multi_send, weak_self, rtp_sender, cb, tracks, ring, poller](uint16_t local_port, const SockException &ex) mutable {
         cb(local_port, ex);
         auto strong_self = weak_self.lock();
         if (!strong_self || ex) {
@@ -328,7 +331,10 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &sender, const MediaSourceE
 
         // 可能归属线程发生变更
         strong_self->getOwnerPoller(MediaSource::NullMediaSource())->async([=]() {
-            strong_self->_rtp_sender[ssrc] = std::move(reader);
+            if(!ssrc_multi_send) {
+                strong_self->_rtp_sender.erase(ssrc);
+            }
+            strong_self->_rtp_sender.emplace(ssrc,reader);
         });
     });
 #else

src/Common/MultiMediaSourceMuxer.h

@@ -173,7 +173,7 @@ private:
     toolkit::Ticker _last_check;
     Stamp _stamp[2];
     std::weak_ptr<Listener> _track_listener;
-    std::unordered_map<std::string, RingType::RingReader::Ptr> _rtp_sender;
+    std::unordered_multimap<std::string, RingType::RingReader::Ptr> _rtp_sender;
     FMP4MediaSourceMuxer::Ptr _fmp4;
     RtmpMediaSourceMuxer::Ptr _rtmp;
     RtspMediaSourceMuxer::Ptr _rtsp;

src/Common/config.cpp

@@ -57,7 +57,7 @@ const string kBroadcastNotFoundStream = "kBroadcastNotFoundStream";
 const string kBroadcastStreamNoneReader = "kBroadcastStreamNoneReader";
 const string kBroadcastHttpBeforeAccess = "kBroadcastHttpBeforeAccess";
 const string kBroadcastSendRtpStopped = "kBroadcastSendRtpStopped";
-const string KBroadcastRtpServerTimeout = "KBroadcastRtpServerTimeout";
+const string kBroadcastRtpServerTimeout = "kBroadcastRtpServerTimeout";
 
 } // namespace Broadcast
 

src/Common/config.h

@@ -106,7 +106,7 @@ extern const std::string kBroadcastReloadConfig;
 #define BroadcastReloadConfigArgs void
 
 // rtp server 超时
-extern const std::string KBroadcastRtpServerTimeout;
+extern const std::string kBroadcastRtpServerTimeout;
 #define BroadcastRtpServerTimeoutArgs uint16_t &local_port, const string &stream_id,int &tcp_mode, bool &re_use_port, uint32_t &ssrc
 
 #define ReloadConfigTag ((void *)(0xFF))

src/Common/strCoding.cpp

@@ -36,20 +36,21 @@ void UnicodeToUTF8(char *pOut, const wchar_t *pText) {
     return;
 }
 
-char CharToInt(char ch) {
+char HexCharToBin(char ch) {
     if (ch >= '0' && ch <= '9') return (char)(ch - '0');
     if (ch >= 'a' && ch <= 'f') return (char)(ch - 'a' + 10);
     if (ch >= 'A' && ch <= 'F') return (char)(ch - 'A' + 10);
     return -1;
 }
 
-char StrToBin(const char *str) {
+char HexStrToBin(const char *str) {
-    char tempWord[2];
+    auto high = HexCharToBin(str[0]);
-    char chn;
+    auto low = HexCharToBin(str[1]);
-    tempWord[0] = CharToInt(str[0]); //make the B to 11 -- 00001011 
+    if (high == -1 || low == -1) {
-    tempWord[1] = CharToInt(str[1]); //make the 0 to 0 -- 00000000 
+        // 无法把16进制字符串转换为二进制
-    chn = (tempWord[0] << 4) | tempWord[1]; //to change the BO to 10110000 
+        return -1;
-    return chn;
+    }
+    return (high << 4) | low;
 }
 
 string strCoding::UrlEncode(const string &str) {
@@ -70,26 +71,51 @@ string strCoding::UrlEncode(const string &str) {
 
 string strCoding::UrlDecode(const string &str) {
     string output;
-    char tmp[2];
     size_t i = 0, len = str.length();
     while (i < len) {
         if (str[i] == '%') {
-            if (i > len - 3) {
+            if (i + 3 > len) {
-                //防止内存溢出
+                // %后面必须还有两个字节才会反转义
+                output.append(str, i, len - i);
                 break;
             }
-            tmp[0] = str[i + 1];
+            char ch = HexStrToBin(&(str[i + 1]));
-            tmp[1] = str[i + 2];
+            if (ch == -1) {
-            output += StrToBin(tmp);
+                // %后面两个字节不是16进制字符串，转义失败；直接拼接3个原始字符
-            i = i + 3;
+                output.append(str, i, 3);
+            } else {
+                output += ch;
+            }
+            i += 3;
         } else {
             output += str[i];
-            i++;
+            ++i;
         }
     }
     return output;
 }
 
+#if 0
+#include "Util/onceToken.h"
+static toolkit::onceToken token([]() {
+    auto str0 = strCoding::UrlDecode(
+        "rtsp%3A%2F%2Fadmin%3AJm13317934%25jm%40111.47.84.69%3A554%2FStreaming%2FChannels%2F101%3Ftransportmode%3Dunicast%26amp%3Bprofile%3DProfile_1");
+    auto str1 = strCoding::UrlDecode("%j1"); // 测试%后面两个字节不是16进制字符串
+    auto str2 = strCoding::UrlDecode("%a"); // 测试%后面字节数不够
+    auto str3 = strCoding::UrlDecode("%"); // 测试只有%
+    auto str4 = strCoding::UrlDecode("%%%"); // 测试多个%
+    auto str5 = strCoding::UrlDecode("%%%%40"); // 测试多个非法%后恢复正常解析
+    auto str6 = strCoding::UrlDecode("Jm13317934%jm"); // 测试多个非法%后恢复正常解析
+    cout << str0 << endl;
+    cout << str1 << endl;
+    cout << str2 << endl;
+    cout << str3 << endl;
+    cout << str4 << endl;
+    cout << str5 << endl;
+    cout << str6 << endl;
+});
+#endif
+
 ///////////////////////////////windows专用///////////////////////////////////
 #if defined(_WIN32)
 void UnicodeToGB2312(char* pOut, wchar_t uData)

src/Extension/AAC.cpp

@@ -67,7 +67,10 @@ static void dumpAdtsHeader(const AdtsHeader &hed, uint8_t *out) {
     out[6] |= (hed.no_raw_data_blocks_in_frame & 0x03); // 2 bit
 }
 
-static void parseAacConfig(const string &config, AdtsHeader &adts) {
+static bool parseAacConfig(const string &config, AdtsHeader &adts) {
+    if (config.size() < 2) {
+        return false;
+    }
     uint8_t cfg1 = config[0];
     uint8_t cfg2 = config[1];
 
@@ -94,6 +97,7 @@ static void parseAacConfig(const string &config, AdtsHeader &adts) {
     adts.aac_frame_length = 7;
     adts.adts_buffer_fullness = 2047;
     adts.no_raw_data_blocks_in_frame = 0;
+    return true;
 }
 #endif// ENABLE_MP4
 
@@ -171,7 +175,9 @@ int dumpAacConfig(const string &config, size_t length, uint8_t *out, size_t out_
 bool parseAacConfig(const string &config, int &samplerate, int &channels) {
 #ifndef ENABLE_MP4
     AdtsHeader header;
-    parseAacConfig(config, header);
+    if (!parseAacConfig(config, header)) {
+        return false;
+    }
     samplerate = samplingFrequencyTable[header.sf_index];
     channels = header.channel_configuration;
     return true;
@@ -326,11 +332,14 @@ bool AACTrack::inputFrame_l(const Frame::Ptr &frame) {
     return false;
 }
 
-void AACTrack::onReady() {
+bool AACTrack::update() {
-    if (_cfg.size() < 2) {
+    return parseAacConfig(_cfg, _sampleRate, _channel);
-        return;
+}
+
+void AACTrack::onReady() {
+    if (!parseAacConfig(_cfg, _sampleRate, _channel)) {
+        _cfg.clear();
     }
-    parseAacConfig(_cfg, _sampleRate, _channel);
 }
 
 Track::Ptr AACTrack::clone() {
@@ -342,6 +351,7 @@ Sdp::Ptr AACTrack::getSdp() {
         WarnL << getCodecName() << " Track未准备好";
         return nullptr;
     }
+    update();
     return std::make_shared<AACSdp>(getConfig(), getAudioSampleRate(), getAudioChannel(), getBitRate() / 1024);
 }
 

src/Extension/AAC.h

@@ -52,6 +52,7 @@ public:
     int getAudioSampleRate() const override;
     int getAudioSampleBit() const override;
     bool inputFrame(const Frame::Ptr &frame) override;
+    bool update() override;
 
 private:
     void onReady();

src/Extension/Frame.h

@@ -16,6 +16,7 @@
 #include <functional>
 #include "Util/List.h"
 #include "Util/TimeTicker.h"
+#include "Common/Stamp.h"
 #include "Network/Buffer.h"
 
 namespace mediakit {
@@ -361,11 +362,18 @@ public:
         return _gop_interval_ms;
     }
 
+    int64_t getDuration() const {
+        std::lock_guard<std::recursive_mutex> lck(_mtx);
+        return _stamp.getRelativeStamp();
+    }
+
 private:
     void doStatistics(const Frame::Ptr &frame) {
         if (!frame->configFrame() && !frame->dropAble()) {
             // 忽略配置帧与可丢弃的帧
             ++_frames;
+            int64_t out;
+            _stamp.revise(frame->dts(), frame->pts(), out, out);
             if (frame->keyFrame() && frame->getTrackType() == TrackVideo) {
                 // 遇视频关键帧时统计
                 ++_video_key_frames;
@@ -384,6 +392,7 @@ private:
     uint64_t _last_frames = 0;
     uint64_t _frames = 0;
     uint64_t _video_key_frames = 0;
+    Stamp _stamp;
     mutable std::recursive_mutex _mtx;
     std::map<void *, FrameWriterInterface::Ptr> _delegates;
 };

src/Extension/H264.cpp

@@ -168,6 +168,10 @@ bool H264Track::inputFrame(const Frame::Ptr &frame) {
     return ret;
 }
 
+bool H264Track::update() {
+    return getAVCInfo(_sps, _width, _height, _fps);
+}
+
 void H264Track::onReady() {
     if (!getAVCInfo(_sps, _width, _height, _fps)) {
         _sps.clear();

src/Extension/H264.h

@@ -128,6 +128,7 @@ public:
     int getVideoWidth() const override;
     float getVideoFps() const override;
     bool inputFrame(const Frame::Ptr &frame) override;
+    bool update() override;
 
 private:
     void onReady();

src/Extension/H264Rtp.cpp

@@ -44,13 +44,15 @@ H264Frame::Ptr H264RtpDecoder::obtainFrame() {
 
 bool H264RtpDecoder::inputRtp(const RtpPacket::Ptr &rtp, bool key_pos) {
     auto seq = rtp->getSeq();
-    auto ret = decodeRtp(rtp);
+    auto last_is_gop = _is_gop;
+    _is_gop = decodeRtp(rtp);
     if (!_gop_dropped && seq != (uint16_t)(_last_seq + 1) && _last_seq) {
         _gop_dropped = true;
         WarnL << "start drop h264 gop, last seq:" << _last_seq << ", rtp:\r\n" << rtp->dumpString();
     }
     _last_seq = seq;
-    return ret;
+    // 确保有sps rtp的时候，gop从sps开始；否则从关键帧开始
+    return _is_gop && !last_is_gop;
 }
 
 /*
@@ -74,7 +76,7 @@ bool H264RtpDecoder::singleFrame(const RtpPacket::Ptr &rtp, const uint8_t *ptr,
     _frame->_buffer.assign("\x00\x00\x00\x01", 4);
     _frame->_buffer.append((char *) ptr, size);
     _frame->_pts = stamp;
-    auto key = _frame->keyFrame();
+    auto key = _frame->keyFrame() || _frame->configFrame();
     outputFrame(rtp, _frame);
     return key;
 }
@@ -127,7 +129,7 @@ bool H264RtpDecoder::mergeFu(const RtpPacket::Ptr &rtp, const uint8_t *ptr, ssiz
 
     if (!fu->end_bit) {
         //非末尾包
-        return fu->start_bit ? _frame->keyFrame() : false;
+        return fu->start_bit ? (_frame->keyFrame() || _frame->configFrame()) : false;
     }
 
     //确保下一次fu必须收到第一个包

src/Extension/H264Rtp.h

@@ -51,6 +51,7 @@ private:
     void outputFrame(const RtpPacket::Ptr &rtp, const H264Frame::Ptr &frame);
 
 private:
+    bool _is_gop = false;
     bool _gop_dropped = false;
     bool _fu_dropped = true;
     uint16_t _last_seq = 0;

src/Extension/H265.cpp

@@ -144,6 +144,10 @@ bool H265Track::inputFrame_l(const Frame::Ptr &frame) {
     return ret;
 }
 
+bool H265Track::update() {
+    return getHEVCInfo(_vps, _sps, _width, _height, _fps);
+}
+
 void H265Track::onReady() {
     if (!getHEVCInfo(_vps, _sps, _width, _height, _fps)) {
         _vps.clear();

src/Extension/H265.h

@@ -150,6 +150,7 @@ public:
     int getVideoHeight() const override;
     float getVideoFps() const override;
     bool inputFrame(const Frame::Ptr &frame) override;
+    bool update() override;
 
 private:
     void onReady();

src/Extension/H265Rtp.cpp

@@ -163,7 +163,7 @@ bool H265RtpDecoder::mergeFu(const RtpPacket::Ptr &rtp, const uint8_t *ptr, ssiz
 
     if (!e_bit) {
         //非末尾包
-        return s_bit ? _frame->keyFrame() : false;
+        return s_bit ? (_frame->keyFrame() || _frame->configFrame()) : false;
     }
 
     //确保下一次fu必须收到第一个包
@@ -175,13 +175,15 @@ bool H265RtpDecoder::mergeFu(const RtpPacket::Ptr &rtp, const uint8_t *ptr, ssiz
 
 bool H265RtpDecoder::inputRtp(const RtpPacket::Ptr &rtp, bool) {
     auto seq = rtp->getSeq();
-    auto ret = decodeRtp(rtp);
+    auto last_is_gop = _is_gop;
+    _is_gop = decodeRtp(rtp);
     if (!_gop_dropped && seq != (uint16_t) (_last_seq + 1) && _last_seq) {
         _gop_dropped = true;
         WarnL << "start drop h265 gop, last seq:" << _last_seq << ", rtp:\r\n" << rtp->dumpString();
     }
     _last_seq = seq;
-    return ret;
+    // 确保有sps rtp的时候，gop从sps开始；否则从关键帧开始
+    return _is_gop && !last_is_gop;
 }
 
 bool H265RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtp) {
@@ -220,7 +222,7 @@ bool H265RtpDecoder::singleFrame(const RtpPacket::Ptr &rtp, const uint8_t *ptr,
     _frame->_buffer.assign("\x00\x00\x00\x01", 4);
     _frame->_buffer.append((char *) ptr, size);
     _frame->_pts = stamp;
-    auto key = _frame->keyFrame();
+    auto key = _frame->keyFrame() || _frame->configFrame();
     outputFrame(rtp, _frame);
     return key;
 }

src/Extension/H265Rtp.h

@@ -51,6 +51,7 @@ private:
     void outputFrame(const RtpPacket::Ptr &rtp, const H265Frame::Ptr &frame);
 
 private:
+    bool _is_gop = false;
     bool _using_donl_field = false;
     bool _gop_dropped = false;
     bool _fu_dropped = true;

src/Extension/Track.h

@@ -39,6 +39,11 @@ public:
      */
     virtual Track::Ptr clone() = 0;
 
+    /**
+     * 更新track信息，比如触发sps/pps解析
+     */
+    virtual bool update() { return false; }
+
     /**
      * 生成sdp
      * @return  sdp对象

src/Http/HttpClient.cpp

@@ -80,7 +80,7 @@ void HttpClient::sendRequest(const string &url) {
     }
 
     if (!alive() || host_changed) {
-        startConnect(host, port, _wait_header_ms);
+        startConnect(host, port, _wait_header_ms / 1000.0f);
     } else {
         SockException ex;
         onConnect_l(ex);

src/Http/HttpSession.cpp

@@ -166,11 +166,11 @@ void HttpSession::onError(const SockException &err) {
     if (_is_live_stream) {
         // flv/ts播放器
         uint64_t duration = _ticker.createdTime() / 1000;
-        WarnP(this) << "FLV/TS/FMP4播放器(" << _mediaInfo.shortUrl() << ")断开:" << err << ",耗时(s):" << duration;
+        WarnP(this) << "FLV/TS/FMP4播放器(" << _media_info.shortUrl() << ")断开:" << err << ",耗时(s):" << duration;
 
         GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
         if (_total_bytes_usage >= iFlowThreshold * 1024) {
-            NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _mediaInfo, _total_bytes_usage, duration, true, *this);
+            NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _media_info, _total_bytes_usage, duration, true, *this);
         }
         return;
     }
@@ -263,9 +263,9 @@ bool HttpSession::checkLiveStream(const string &schema, const string &url_suffix
     }
 
     // 解析带上协议+参数完整的url
-    _mediaInfo.parse(schema + "://" + _parser["Host"] + url);
+    _media_info.parse(schema + "://" + _parser["Host"] + url);
 
-    if (_mediaInfo.app.empty() || _mediaInfo.stream.empty()) {
+    if (_media_info.app.empty() || _media_info.stream.empty()) {
         // url不合法
         return false;
     }
@@ -288,7 +288,7 @@ bool HttpSession::checkLiveStream(const string &schema, const string &url_suffix
         }
 
         // 异步查找直播流
-        MediaSource::findAsync(strong_self->_mediaInfo, strong_self, [weak_self, close_flag, cb](const MediaSource::Ptr &src) {
+        MediaSource::findAsync(strong_self->_media_info, strong_self, [weak_self, close_flag, cb](const MediaSource::Ptr &src) {
             auto strong_self = weak_self.lock();
             if (!strong_self) {
                 // 本对象已经销毁
@@ -311,7 +311,7 @@ bool HttpSession::checkLiveStream(const string &schema, const string &url_suffix
         }
     };
 
-    auto flag = NOTICE_EMIT(BroadcastMediaPlayedArgs, Broadcast::kBroadcastMediaPlayed, _mediaInfo, invoker, *this);
+    auto flag = NOTICE_EMIT(BroadcastMediaPlayedArgs, Broadcast::kBroadcastMediaPlayed, _media_info, invoker, *this);
     if (!flag) {
         // 该事件无人监听,默认不鉴权
         onRes("");

src/Http/HttpSession.h

@@ -124,7 +124,7 @@ private:
     void setSocketFlags();
 
 protected:
-    MediaInfo _mediaInfo;
+    MediaInfo _media_info;
 
 private:
     bool _is_live_stream = false;

src/Http/WebSocketSession.h

@@ -69,7 +69,7 @@ template <typename SessionType>
 class SessionCreator {
 public:
     //返回的Session必须派生于SendInterceptor，可以返回null
-    toolkit::Session::Ptr operator()(const mediakit::Parser &header, const mediakit::HttpSession &parent, const toolkit::Socket::Ptr &pSock){
+    toolkit::Session::Ptr operator()(const mediakit::Parser &header, const mediakit::HttpSession &parent, const toolkit::Socket::Ptr &pSock, mediakit::WebSocketHeader::Type &data_type){
         return std::make_shared<SessionTypeImp<SessionType> >(header,parent,pSock);
     }
 };
@@ -128,7 +128,8 @@ protected:
      */
     bool onWebSocketConnect(const mediakit::Parser &header) override{
         //创建websocket session类
-        _session = _creator(header, *this, HttpSessionType::getSock());
+        auto data_type = DataType;
+        _session = _creator(header, *this, HttpSessionType::getSock(), data_type);
         if (!_session) {
             // 此url不允许创建websocket连接
             return false;
@@ -140,13 +141,13 @@ protected:
 
         //此处截取数据并进行websocket协议打包
         std::weak_ptr<WebSocketSessionBase> weakSelf = std::static_pointer_cast<WebSocketSessionBase>(HttpSessionType::shared_from_this());
-        std::dynamic_pointer_cast<SendInterceptor>(_session)->setOnBeforeSendCB([weakSelf](const toolkit::Buffer::Ptr &buf) {
+        std::dynamic_pointer_cast<SendInterceptor>(_session)->setOnBeforeSendCB([weakSelf, data_type](const toolkit::Buffer::Ptr &buf) {
             auto strongSelf = weakSelf.lock();
             if (strongSelf) {
                 mediakit::WebSocketHeader header;
                 header._fin = true;
                 header._reserved = 0;
-                header._opcode = DataType;
+                header._opcode = data_type;
                 header._mask_flag = false;
                 strongSelf->HttpSessionType::encode(header, buf);
             }

src/Player/PlayerProxy.cpp

@@ -70,6 +70,7 @@ void PlayerProxy::setTranslationInfo()
     _transtalion_info.stream_info.clear();
     auto tracks = _muxer->getTracks();
     for (auto &track : tracks) {
+        track->update();
         _transtalion_info.stream_info.emplace_back();
         auto &back = _transtalion_info.stream_info.back();
         back.bitrate = track->getBitRate();

src/Record/MP4Demuxer.cpp

@@ -16,8 +16,10 @@
 #include "Extension/AAC.h"
 #include "Extension/G711.h"
 #include "Extension/Opus.h"
-using namespace toolkit;
+#include "Extension/JPEG.h"
+
 using namespace std;
+using namespace toolkit;
 
 namespace mediakit {
 
@@ -105,8 +107,9 @@ void MP4Demuxer::onVideoTrack(uint32_t track, uint8_t object, int width, int hei
                     video->inputFrame(std::make_shared<H264FrameNoCacheAble>((char *)config, size, 0, 0,4));
                 }
             }
-        }
             break;
+        }
+
         case MOV_OBJECT_HEVC: {
             auto video = std::make_shared<H265Track>();
             _track_to_codec.emplace(track,video);
@@ -120,11 +123,16 @@ void MP4Demuxer::onVideoTrack(uint32_t track, uint8_t object, int width, int hei
                     video->inputFrame(std::make_shared<H265FrameNoCacheAble>((char *) config, size, 0, 0,4));
                 }
             }
+            break;
         }
+
+        case MOV_OBJECT_JPEG: {
+            auto video = std::make_shared<JPEGTrack>();
+            _track_to_codec.emplace(track,video);
             break;
-        default:
+        }
-            WarnL << "不支持该编码类型的MP4,已忽略:" << getObjectName(object);
+
-            break;
+        default: WarnL << "不支持该编码类型的MP4,已忽略:" << getObjectName(object); break;
     }
 }
 
@@ -243,6 +251,11 @@ Frame::Ptr MP4Demuxer::makeFrame(uint32_t track_id, const Buffer::Ptr &buf, int6
             break;
         }
 
+        case CodecJPEG: {
+            ret = std::make_shared<JPEGFrame>(buf, (uint64_t)dts, 0, DATA_OFFSET);
+            break;
+        }
+
         case CodecAAC: {
             AACTrack::Ptr track = dynamic_pointer_cast<AACTrack>(it->second);
             assert(track);

src/Record/MP4Muxer.cpp

@@ -198,6 +198,7 @@ bool MP4MuxerInterface::addTrack(const Track::Ptr &track) {
         return false;
     }
 
+    track->update();
     switch (track->getCodecId()) {
         case CodecG711A:
         case CodecG711U:

src/Record/MP4Recorder.cpp

@@ -34,8 +34,12 @@ MP4Recorder::MP4Recorder(const string &path, const string &vhost, const string &
 }
 
 MP4Recorder::~MP4Recorder() {
+    try {
         flush();
         closeFile();
+    } catch (std::exception &ex) {
+        WarnL << ex.what();
+    }
 }
 
 void MP4Recorder::createFile() {

src/Rtmp/Rtmp.cpp

@@ -57,6 +57,7 @@ AudioMeta::AudioMeta(const AudioTrack::Ptr &audio) {
 }
 
 uint8_t getAudioRtmpFlags(const Track::Ptr &track) {
+    track->update();
     switch (track->getTrackType()) {
         case TrackAudio: {
             auto audioTrack = std::dynamic_pointer_cast<AudioTrack>(track);
@@ -115,6 +116,7 @@ uint8_t getAudioRtmpFlags(const Track::Ptr &track) {
 
 void Metadata::addTrack(AMFValue &metadata, const Track::Ptr &track) {
     Metadata::Ptr new_metadata;
+    track->update();
     switch (track->getTrackType()) {
         case TrackVideo: {
             new_metadata = std::make_shared<VideoMeta>(std::dynamic_pointer_cast<VideoTrack>(track));

src/Rtmp/RtmpPlayer.cpp

@@ -52,17 +52,18 @@ void RtmpPlayer::teardown() {
 
 void RtmpPlayer::play(const string &url)  {
     teardown();
-    string host_url = findSubString(url.data(), "://", "/");
+    auto schema = findSubString(url.data(), nullptr, "://");
-    {
+    auto host_url = findSubString(url.data(), "://", "/");
-        auto pos = url.find_last_of('/');
+    _app = findSubString(url.data(), (host_url + "/").data(), "/");
-        if (pos != string::npos) {
+    _stream_id = findSubString(url.data(), (host_url + "/" + _app + "/").data(), NULL);
-            _stream_id = url.substr(pos + 1);
+    auto app_second = findSubString(_stream_id.data(), nullptr, "/");
+    if (!app_second.empty() && app_second.find('?') == std::string::npos) {
+        // _stream_id存在多级；不包含'?', 说明分割符'/'不是url参数的一部分
+        _app += "/" + app_second;
+        _stream_id.erase(0, app_second.size() + 1);
     }
-    }
+    _tc_url = schema + "://" + host_url + "/" + _app;
-    _app = findSubString(url.data(), (host_url + "/").data(), ("/" + _stream_id).data());
+    if (_app.empty() || _stream_id.empty()) {
-    _tc_url = string("rtmp://") + host_url + "/" + _app;
-
-    if (!_app.size() || !_stream_id.size()) {
         onPlayResult_l(SockException(Err_other, "rtmp url非法"), false);
         return;
     }

src/Rtmp/RtmpPusher.cpp

@@ -65,12 +65,18 @@ void RtmpPusher::onPublishResult_l(const SockException &ex, bool handshake_done)
 
 void RtmpPusher::publish(const string &url) {
     teardown();
-    string host_url = findSubString(url.data(), "://", "/");
+    auto schema = findSubString(url.data(), nullptr, "://");
+    auto host_url = findSubString(url.data(), "://", "/");
     _app = findSubString(url.data(), (host_url + "/").data(), "/");
     _stream_id = findSubString(url.data(), (host_url + "/" + _app + "/").data(), NULL);
-    _tc_url = string("rtmp://") + host_url + "/" + _app;
+    auto app_second = findSubString(_stream_id.data(), nullptr, "/");
-
+    if (!app_second.empty() && app_second.find('?') == std::string::npos) {
-    if (!_app.size() || !_stream_id.size()) {
+        // _stream_id存在多级；不包含'?', 说明分割符'/'不是url参数的一部分
+        _app += "/" + app_second;
+        _stream_id.erase(0, app_second.size() + 1);
+    }
+    _tc_url = schema + "://" + host_url + "/" + _app;
+    if (_app.empty() || _stream_id.empty()) {
         onPublishResult_l(SockException(Err_other, "rtmp url非法"), false);
         return;
     }

src/Rtp/RtpSender.cpp

@@ -140,7 +140,7 @@ void RtpSender::startSend(const MediaSourceEvent::SendRtpArgs &args, const funct
                     cb(0, SockException(Err_other, ex.what()));
                     return;
                 }
-                strong_self->_socket_rtp->bindPeerAddr((struct sockaddr *)&addr);
+                strong_self->_socket_rtp->bindPeerAddr((struct sockaddr *)&addr, 0, true);
                 strong_self->onConnect();
                 cb(strong_self->_socket_rtp->get_local_port(), SockException());
             });
@@ -182,7 +182,7 @@ void RtpSender::createRtcpSocket() {
         case AF_INET6: ((sockaddr_in6 *)&addr)->sin6_port = htons(ntohs(((sockaddr_in6 *)&addr)->sin6_port) + 1); break;
         default: assert(0); break;
     }
-    _socket_rtcp->bindPeerAddr((struct sockaddr *)&addr);
+    _socket_rtcp->bindPeerAddr((struct sockaddr *)&addr, 0, true);
 
     _rtcp_context = std::make_shared<RtcpContextForSend>();
     weak_ptr<RtpSender> weak_self = shared_from_this();

src/Rtp/RtpServer.cpp

@@ -89,6 +89,8 @@ public:
             for (auto &rtcp : rtcps) {
                 strong_self->_process->onRtcp(rtcp);
             }
+            // 收到sr rtcp后驱动返回rr rtcp
+            strong_self->sendRtcp(strong_self->_ssrc, (struct sockaddr *)(strong_self->_rtcp_addr.get()));
         });
 
         GET_CONFIG(uint64_t, timeoutSec, RtpProxy::kTimeoutSec);
@@ -102,7 +104,7 @@ public:
                     process->setOnDetach(std::move(strong_self->_on_detach));
                 }
                 if (!process) { // process 未创建，触发rtp server 超时事件
-                    NOTICE_EMIT(BroadcastRtpServerTimeoutArgs, Broadcast::KBroadcastRtpServerTimeout, strong_self->_local_port, strong_self->_stream_id,
+                    NOTICE_EMIT(BroadcastRtpServerTimeoutArgs, Broadcast::kBroadcastRtpServerTimeout, strong_self->_local_port, strong_self->_stream_id,
                                 (int)strong_self->_tcp_mode, strong_self->_re_use_port, strong_self->_ssrc);
                 }
             }
@@ -154,7 +156,7 @@ private:
     EventPoller::DelayTask::Ptr _delay_task;
 };
 
-void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, bool only_audio) {
+void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, bool only_audio, bool multiplex) {
     //创建udp服务器
     Socket::Ptr rtp_socket = Socket::createSocket(nullptr, true);
     Socket::Ptr rtcp_socket = Socket::createSocket(nullptr, true);
@@ -193,7 +195,8 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
     //创建udp服务器
     UdpServer::Ptr udp_server;
     RtcpHelper::Ptr helper;
-    if (!stream_id.empty()) {
+    //增加了多路复用判断，如果多路复用为true，就走else逻辑，同时保留了原来stream_id为空走else逻辑
+    if (!stream_id.empty() && !multiplex) {
         //指定了流id，那么一个端口一个流(不管是否包含多个ssrc的多个流，绑定rtp源后，会筛选掉ip端口不匹配的流)
         helper = std::make_shared<RtcpHelper>(std::move(rtcp_socket), stream_id);
         helper->startRtcp();

src/Rtp/RtpServer.h

@@ -42,9 +42,10 @@ public:
      * @param local_ip 绑定的本地网卡ip
      * @param re_use_port 是否设置socket为re_use属性
      * @param ssrc 指定的ssrc
+     * @param multiplex 多路复用
      */
     void start(uint16_t local_port, const std::string &stream_id = "", TcpMode tcp_mode = PASSIVE,
-               const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0, bool only_audio = false);
+               const char *local_ip = "::", bool re_use_port = true, uint32_t ssrc = 0, bool only_audio = false, bool multiplex = false);
 
     /**
      * 连接到tcp服务(tcp主动模式)

src/TS/TSMediaSourceMuxer.h

@@ -26,7 +26,13 @@ public:
         _media_src = std::make_shared<TSMediaSource>(tuple);
     }
 
-    ~TSMediaSourceMuxer() override { MpegMuxer::flush(); };
+    ~TSMediaSourceMuxer() override {
+        try {
+            MpegMuxer::flush();
+        } catch (std::exception &ex) {
+            WarnL << ex.what();
+        }
+    };
 
     void setListener(const std::weak_ptr<MediaSourceEvent> &listener){
         setDelegate(listener);

tests/test_wsServer.cpp

@@ -81,9 +81,11 @@ public:
  */
 struct EchoSessionCreator {
     //返回的Session必须派生于SendInterceptor，可以返回null(拒绝连接)
-    Session::Ptr operator()(const Parser &header, const HttpSession &parent, const Socket::Ptr &pSock) {
+    Session::Ptr operator()(const Parser &header, const HttpSession &parent, const Socket::Ptr &pSock, mediakit::WebSocketHeader::Type &type) {
 //        return nullptr;
         if (header.url() == "/") {
+            // 可以指定传输方式
+            // type = mediakit::WebSocketHeader::BINARY;
             return std::make_shared<SessionTypeImp<EchoSession> >(header, parent, pSock);
         }
         return std::make_shared<SessionTypeImp<EchoSessionWithUrl> >(header, parent, pSock);

webrtc/DtlsTransport.cpp

@@ -29,6 +29,8 @@ OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
 #include <cstdio>  // std::sprintf(), std::fopen()
 #include <cstring> // std::memcpy(), std::strcmp()
 #include "Util/util.h"
+#include "Util/SSLBox.h"
+#include "Util/SSLUtil.h"
 
 using namespace std;
 
@@ -129,16 +131,10 @@ namespace RTC
         MS_TRACE();
 
         // Generate a X509 certificate and private key (unless PEM files are provided).
-        if (true /*
+        auto ssl = toolkit::SSL_Initor::Instance().getSSLCtx("", true);
-          Settings::configuration.dtlsCertificateFile.empty() ||
+        if (!ssl || !ReadCertificateAndPrivateKeyFromContext(ssl.get())) {
-          Settings::configuration.dtlsPrivateKeyFile.empty()*/)
-        {
             GenerateCertificateAndPrivateKey();
         }
-        else
-        {
-            ReadCertificateAndPrivateKeyFromFiles();
-        }
 
         // Create a global SSL_CTX.
         CreateSslCtx();
@@ -297,59 +293,22 @@ namespace RTC
         MS_THROW_ERROR("DTLS certificate and private key generation failed");
     }
 
-    void DtlsTransport::DtlsEnvironment::ReadCertificateAndPrivateKeyFromFiles()
+    bool DtlsTransport::DtlsEnvironment::ReadCertificateAndPrivateKeyFromContext(SSL_CTX *ctx)
     {
-#if 0
         MS_TRACE();
-
+        certificate = SSL_CTX_get0_certificate(ctx);
-        FILE* file{ nullptr };
+        if (!certificate) {
-
+            return false;
-        file = fopen(Settings::configuration.dtlsCertificateFile.c_str(), "r");
-
-        if (!file)
-        {
-            MS_ERROR("error reading DTLS certificate file: %s", std::strerror(errno));
-
-            goto error;
         }
+        X509_up_ref(certificate);
 
-        certificate = PEM_read_X509(file, nullptr, nullptr, nullptr);
+        privateKey = SSL_CTX_get0_privatekey(ctx);
-
+        if (!privateKey) {
-        if (!certificate)
+            return false;
-        {
-            LOG_OPENSSL_ERROR("PEM_read_X509() failed");
-
-            goto error;
         }
-
+        EVP_PKEY_up_ref(privateKey);
-        fclose(file);
+        InfoL << "Load webrtc dtls certificate: " << toolkit::SSLUtil::getServerName(certificate);
-
+        return true;
-        file = fopen(Settings::configuration.dtlsPrivateKeyFile.c_str(), "r");
-
-        if (!file)
-        {
-            MS_ERROR("error reading DTLS private key file: %s", std::strerror(errno));
-
-            goto error;
-        }
-
-        privateKey = PEM_read_PrivateKey(file, nullptr, nullptr, nullptr);
-
-        if (!privateKey)
-        {
-            LOG_OPENSSL_ERROR("PEM_read_PrivateKey() failed");
-
-            goto error;
-        }
-
-        fclose(file);
-
-        return;
-
-    error:
-
-        MS_THROW_ERROR("error reading DTLS certificate and private key PEM files");
-#endif
     }
 
     void DtlsTransport::DtlsEnvironment::CreateSslCtx()

webrtc/DtlsTransport.hpp

@@ -88,7 +88,7 @@ namespace RTC
         private:
             DtlsEnvironment();
             void GenerateCertificateAndPrivateKey();
-            void ReadCertificateAndPrivateKeyFromFiles();
+            bool ReadCertificateAndPrivateKeyFromContext(SSL_CTX *ctx);
             void CreateSslCtx();
             void GenerateFingerprints();
 

webrtc/WebRtcTransport.cpp

@@ -251,7 +251,7 @@ void WebRtcTransport::sendSockData(const char *buf, size_t len, RTC::TransportTu
 }
 
 Session::Ptr WebRtcTransport::getSession() const {
-    auto tuple = _ice_server->GetSelectedTuple(true);
+    auto tuple = _ice_server ? _ice_server->GetSelectedTuple(true) : nullptr;
     return tuple ? static_pointer_cast<Session>(tuple->shared_from_this()) : nullptr;
 }
 

www/webassist

@@ -0,0 +1 @@
+Subproject commit b02d2a4c1abf95db45e50bb77d789defa0fcc4b7