sync from master

AUTHORS

@@ -44,7 +44,6 @@ Xinghua Zhao <(holychaossword@hotmail.com>
 [Dw9](https://github.com/Dw9)
 明月惊鹊 <mingyuejingque@gmail.com>
 cgm <2958580318@qq.com>
-hejilin <1724010622@qq.com>
 alexliyu7352 <liyu7352@gmail.com>
 cgm <2958580318@qq.com>
 [haorui wang](https://github.com/HaoruiWang)
@@ -104,3 +103,8 @@ WuPeng <wp@zafu.edu.cn>
 [sandro-qiang](https://github.com/sandro-qiang)
 [Paul Philippov](https://github.com/themactep)
 [张传峰](https://github.com/zhang-chuanfeng)
+[lidaofu-hub](https://github.com/lidaofu-hub)
+[huangcaichun](https://github.com/huangcaichun)
+[jamesZHANG500](https://github.com/jamesZHANG500)
+[weidelong](https://github.com/wdl1697454803)
+[小强先生](https://github.com/linshangqiang)

CMakeLists.txt

@@ -57,7 +57,8 @@ option(ENABLE_SCTP "Enable SCTP" ON)
 option(ENABLE_WEBRTC "Enable WebRTC" ON)
 option(ENABLE_X264 "Enable x264" OFF)
 option(ENABLE_WEPOLL "Enable wepoll" ON)
-option(DISABLE_REPORT "Disable report to report.zlmediakit.com" off)
+option(ENABLE_VIDEOSTACK "Enable video stack" OFF)
+option(DISABLE_REPORT "Disable report to report.zlmediakit.com" OFF)
 option(USE_SOLUTION_FOLDERS "Enable solution dir supported" ON)
 ##############################################################################
 # 设置socket默认缓冲区大小为256k.如果设置为0则不设置socket的默认缓冲区大小,使用系统内核默认值(设置为0仅对linux有效)
@@ -141,8 +142,8 @@ if(GIT_FOUND)
 endif()
 
 configure_file(
-  ${CMAKE_CURRENT_SOURCE_DIR}/version.h.ini
+  ${CMAKE_CURRENT_SOURCE_DIR}/ZLMVersion.h.ini
-  ${CMAKE_CURRENT_BINARY_DIR}/version.h
+  ${CMAKE_CURRENT_BINARY_DIR}/ZLMVersion.h
   @ONLY)
 
 message(STATUS "Git version is ${BRANCH_NAME} ${COMMIT_HASH}/${COMMIT_TIME} ${BUILD_TIME}")
@@ -191,7 +192,7 @@ if(UNIX)
   set(COMPILE_OPTIONS_DEFAULT
     "-fPIC"
     "-Wall;-Wextra"
-    "-Wno-unused-function;-Wno-unused-parameter;-Wno-unused-variable"
+    "-Wno-unused-function;-Wno-unused-parameter;-Wno-unused-variable;-Wno-deprecated-declarations"
     "-Wno-error=extra;-Wno-error=missing-field-initializers;-Wno-error=type-limits")
 
   if("${CMAKE_BUILD_TYPE}" STREQUAL "Debug")
@@ -535,6 +536,6 @@ file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/default.pem" DESTINATION ${EXECUTABLE_OUT
 
 # 拷贝VideoStack 无视频流时默认填充的背景图片
 # Copy the default background image used by VideoStack when there is no video stream
-if (ENABLE_FFMPEG AND ENABLE_X264)
+if (ENABLE_VIDEOSTACK AND ENABLE_FFMPEG AND ENABLE_X264)
   file(COPY "${CMAKE_CURRENT_SOURCE_DIR}/conf/novideo.yuv" DESTINATION ${EXECUTABLE_OUTPUT_PATH})
 endif ()

README.md

@@ -358,6 +358,11 @@ bash build_docker_images.sh
 [sandro-qiang](https://github.com/sandro-qiang)
 [Paul Philippov](https://github.com/themactep)
 [张传峰](https://github.com/zhang-chuanfeng)
+[lidaofu-hub](https://github.com/lidaofu-hub)
+[huangcaichun](https://github.com/huangcaichun)
+[jamesZHANG500](https://github.com/jamesZHANG500)
+[weidelong](https://github.com/wdl1697454803)
+[小强先生](https://github.com/linshangqiang)
 
 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

@@ -516,6 +516,11 @@ Thanks to all those who have supported this project in various ways, including b
 [sandro-qiang](https://github.com/sandro-qiang)
 [Paul Philippov](https://github.com/themactep)
 [张传峰](https://github.com/zhang-chuanfeng)
+[lidaofu-hub](https://github.com/lidaofu-hub)
+[huangcaichun](https://github.com/huangcaichun)
+[jamesZHANG500](https://github.com/jamesZHANG500)
+[weidelong](https://github.com/wdl1697454803)
+[小强先生](https://github.com/linshangqiang)
 
 Also thank to JetBrains for their support for open source project, we developed and debugged zlmediakit with CLion:
 

api/include/mk_common.h

@@ -24,6 +24,10 @@
 #    define API_CALL
 #endif
 
+#ifndef _WIN32
+#define _strdup strdup
+#endif
+
 #if defined(_WIN32) && defined(_MSC_VER)
 #    if !defined(GENERATE_EXPORT)
 #        if defined(MediaKitApi_EXPORTS)

api/include/mk_events_objects.h

@@ -103,6 +103,16 @@ API_EXPORT int API_CALL mk_media_source_get_track_count(const mk_media_source ct
 API_EXPORT mk_track API_CALL mk_media_source_get_track(const mk_media_source ctx, int index);
 // MediaSource::broadcastMessage
 API_EXPORT int API_CALL mk_media_source_broadcast_msg(const mk_media_source ctx, const char *msg, size_t len);
+// MediaSource::getOriginUrl()
+API_EXPORT const char* API_CALL mk_media_source_get_origin_url(const mk_media_source ctx);
+// MediaSource::getOriginType()
+API_EXPORT int API_CALL mk_media_source_get_origin_type(const mk_media_source ctx);
+// MediaSource::getCreateStamp()
+API_EXPORT uint64_t API_CALL mk_media_source_get_create_stamp(const mk_media_source ctx);
+// MediaSource::isRecording()  0:hls,1:MP4
+API_EXPORT int API_CALL mk_media_source_is_recording(const mk_media_source ctx, int type);
+
+
 
 /**
  * 直播源在ZLMediaKit中被称作为MediaSource，
@@ -142,7 +152,7 @@ API_EXPORT void API_CALL mk_media_source_find(const char *schema,
                                               void *user_data,
                                               on_mk_media_source_find_cb cb);
 
-API_EXPORT const mk_media_source API_CALL mk_media_source_find2(const char *schema,
+API_EXPORT mk_media_source API_CALL mk_media_source_find2(const char *schema,
                                                           const char *vhost,
                                                           const char *app,
                                                           const char *stream,

api/source/mk_common.cpp

@@ -304,10 +304,10 @@ API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data
     std::string offer_str = offer;
     std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
     auto args = std::make_shared<WebRtcArgsUrl>(url);
-    WebRtcPluginManager::Instance().getAnswerSdp(*session, type, *args,
+    WebRtcPluginManager::Instance().negotiateSdp(*session, type, *args, [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
-                                                 [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
+        auto &handler = const_cast<WebRtcInterface &>(exchanger);
         try {
-            auto sdp_answer = exchangeSdp(exchanger, offer_str);
+            auto sdp_answer = handler.getAnswerSdp(offer_str);
             cb(ptr.get(), sdp_answer.data(), nullptr);
         } catch (std::exception &ex) {
             cb(ptr.get(), nullptr, ex.what());

api/source/mk_events.cpp

@@ -14,7 +14,10 @@
 #include "Http/HttpSession.h"
 #include "Rtsp/RtspSession.h"
 #include "Record/MP4Recorder.h"
+
+#ifdef ENABLE_WEBRTC
 #include "webrtc/WebRtcTransport.h"
+#endif
 
 using namespace toolkit;
 using namespace mediakit;
@@ -168,7 +171,7 @@ API_EXPORT void API_CALL mk_events_listen(const mk_events *events){
                                                    sender.getMediaTuple().stream.c_str(), ssrc.c_str(), ex.getErrCode(), ex.what());
             }
         });
-
+#ifdef ENABLE_WEBRTC
         NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpConnecting,[](BroadcastRtcSctpConnectArgs){
             if (s_events.on_mk_rtc_sctp_connecting) {
                 s_events.on_mk_rtc_sctp_connecting((mk_rtc_transport)&sender);
@@ -204,6 +207,7 @@ API_EXPORT void API_CALL mk_events_listen(const mk_events *events){
                 s_events.on_mk_rtc_sctp_received((mk_rtc_transport)&sender, streamId, ppid, msg, len);
             }
         });
+#endif
     });
 
 }

api/source/mk_events_objects.cpp

@@ -17,7 +17,10 @@
 
 #include "Http/HttpClient.h"
 #include "Rtsp/RtspSession.h"
+
+#ifdef ENABLE_WEBRTC
 #include "webrtc/WebRtcTransport.h"
+#endif
 
 using namespace toolkit;
 using namespace mediakit;
@@ -127,7 +130,7 @@ API_EXPORT const char* API_CALL mk_parser_get_content(const mk_parser ctx, size_
 API_EXPORT const char* API_CALL mk_media_info_get_params(const mk_media_info ctx){
     assert(ctx);
     MediaInfo *info = (MediaInfo *)ctx;
-    return info->param_strs.c_str();
+    return info->params.c_str();
 }
 
 API_EXPORT const char* API_CALL mk_media_info_get_schema(const mk_media_info ctx){
@@ -225,6 +228,30 @@ API_EXPORT int API_CALL mk_media_source_broadcast_msg(const mk_media_source ctx,
     return src->broadcastMessage(any);
 }
 
+API_EXPORT const char* API_CALL mk_media_source_get_origin_url(const mk_media_source ctx) {
+    assert(ctx);
+    MediaSource *src = (MediaSource *)ctx;
+    return _strdup(src->getOriginUrl().c_str());
+}
+
+API_EXPORT  int API_CALL mk_media_source_get_origin_type(const mk_media_source ctx) {
+    assert(ctx);
+    MediaSource *src = (MediaSource *)ctx;
+    return static_cast<int>(src->getOriginType());
+}
+
+API_EXPORT uint64_t API_CALL mk_media_source_get_create_stamp(const mk_media_source ctx) {
+    assert(ctx);
+    MediaSource *src = (MediaSource *)ctx;
+    return src->getCreateStamp();
+}
+
+API_EXPORT int API_CALL mk_media_source_is_recording(const mk_media_source ctx,int type) {
+    assert(ctx);
+    MediaSource *src = (MediaSource *)ctx;
+    return src->isRecording((Recorder::type)type);
+}
+
 API_EXPORT int API_CALL mk_media_source_close(const mk_media_source ctx,int force){
     assert(ctx);
     MediaSource *src = (MediaSource *)ctx;
@@ -275,7 +302,7 @@ API_EXPORT void API_CALL mk_media_source_find(const char *schema,
     cb(user_data, (mk_media_source)src.get());
 }
 
-API_EXPORT const mk_media_source API_CALL mk_media_source_find2(const char *schema,
+API_EXPORT mk_media_source API_CALL mk_media_source_find2(const char *schema,
                                                           const char *vhost,
                                                           const char *app,
                                                           const char *stream,

api/source/mk_util.cpp

@@ -21,10 +21,6 @@ using namespace std;
 using namespace toolkit;
 using namespace mediakit;
 
-#ifndef _WIN32
-#define _strdup strdup
-#endif
-
 API_EXPORT void API_CALL mk_free(void *ptr) {
     free(ptr);
 }

api/tests/h264_pusher.c

@@ -0,0 +1,152 @@
+/*
+ * Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
+ *
+ * This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
+ *
+ * Use of this source code is governed by MIT-like license that can be found in the
+ * LICENSE file in the root of the source tree. All contributing project authors
+ * may be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <signal.h>
+#include <stdio.h>
+#include <string.h>
+#ifdef _WIN32
+#include "windows.h"
+#else
+#include "unistd.h"
+#endif
+#include "mk_mediakit.h"
+
+static int exit_flag = 0;
+static void s_on_exit(int sig) {
+    exit_flag = 1;
+}
+
+static void on_h264_frame(void *user_data, mk_h264_splitter splitter, const char *data, int size) {
+#ifdef _WIN32
+    Sleep(40);
+#else
+    usleep(40 * 1000);
+#endif
+    static int dts = 0;
+    mk_frame frame = mk_frame_create(MKCodecH264, dts, dts, data, size, NULL, NULL);
+    dts += 40;
+    mk_media_input_frame((mk_media)user_data, frame);
+    mk_frame_unref(frame);
+}
+
+typedef struct {
+    mk_pusher pusher;
+    char *url;
+} Context;
+
+void release_context(void *user_data) {
+    Context *ptr = (Context *)user_data;
+    if (ptr->pusher) {
+        mk_pusher_release(ptr->pusher);
+    }
+    free(ptr->url);
+    free(ptr);
+    log_info("停止推流");
+}
+
+void on_push_result(void *user_data, int err_code, const char *err_msg) {
+    Context *ptr = (Context *)user_data;
+    if (err_code == 0) {
+        log_info("推流成功: %s", ptr->url);
+    } else {
+        log_warn("推流%s失败: %d(%s)", ptr->url, err_code, err_msg);
+    }
+}
+
+void on_push_shutdown(void *user_data, int err_code, const char *err_msg) {
+    Context *ptr = (Context *)user_data;
+    log_warn("推流%s中断: %d(%s)", ptr->url, err_code, err_msg);
+}
+
+void API_CALL on_regist(void *user_data, mk_media_source sender, int regist) {
+    Context *ptr = (Context *)user_data;
+    const char *schema = mk_media_source_get_schema(sender);
+    if (strstr(ptr->url, schema) != ptr->url) {
+        // 协议匹配失败
+        return;
+    }
+
+    if (!regist) {
+        // 注销
+        if (ptr->pusher) {
+            mk_pusher_release(ptr->pusher);
+            ptr->pusher = NULL;
+        }
+    } else {
+        // 注册
+        if (!ptr->pusher) {
+            ptr->pusher = mk_pusher_create_src(sender);
+            mk_pusher_set_on_result2(ptr->pusher, on_push_result, ptr, NULL);
+            mk_pusher_set_on_shutdown2(ptr->pusher, on_push_shutdown, ptr, NULL);
+            // 开始推流
+            mk_pusher_publish(ptr->pusher, ptr->url);
+        }
+    }
+}
+
+int main(int argc, char *argv[]) {
+    if (argc < 3) {
+        log_error("Usage: /path/to/h264/file rtsp_or_rtmp_url");
+        return -1;
+    }
+    mk_config config = { .ini = NULL,
+                         .ini_is_path = 1,
+                         .log_level = 0,
+                         .log_mask = LOG_CONSOLE,
+                         .log_file_path = NULL,
+                         .log_file_days = 0,
+                         .ssl = NULL,
+                         .ssl_is_path = 1,
+                         .ssl_pwd = NULL,
+                         .thread_num = 0 };
+    mk_env_init(&config);
+
+    FILE *fp = fopen(argv[1], "rb");
+    if (!fp) {
+        log_error("打开文件失败!");
+        return -1;
+    }
+
+    mk_media media = mk_media_create("__defaultVhost__", "live", "test", 0, 0, 0);
+    // h264的codec
+    codec_args v_args = { 0 };
+    mk_track v_track = mk_track_create(MKCodecH264, &v_args);
+    mk_media_init_track(media, v_track);
+    mk_media_init_complete(media);
+    mk_track_unref(v_track);
+
+    Context *ctx = (Context *)malloc(sizeof(Context));
+    memset(ctx, 0, sizeof(Context));
+    ctx->url = strdup(argv[2]);
+
+    mk_media_set_on_regist2(media, on_regist, ctx, release_context);
+
+    // 创建h264分帧器
+    mk_h264_splitter splitter = mk_h264_splitter_create(on_h264_frame, media, 0);
+    signal(SIGINT, s_on_exit); // 设置退出信号
+    signal(SIGTERM, s_on_exit); // 设置退出信号
+
+    char buf[1024];
+    while (!exit_flag) {
+        int size = fread(buf, 1, sizeof(buf) - 1, fp);
+        if (size > 0) {
+            mk_h264_splitter_input_data(splitter, buf, size);
+        } else {
+            // 文件读完了，重新开始
+            fseek(fp, 0, SEEK_SET);
+        }
+    }
+
+    log_info("文件读取完毕");
+    mk_h264_splitter_release(splitter);
+    mk_media_release(media);
+    fclose(fp);
+    return 0;
+}

conf/config.ini

@@ -357,7 +357,7 @@ tcpPort = 8000
 rembBitRate=0
 #rtc支持的音频codec类型,在前面的优先级更高
 #以下范例为所有支持的音频codec
-preferredCodecA=PCMU,PCMA,opus,mpeg4-generic
+preferredCodecA=PCMA,PCMU,opus,mpeg4-generic
 #rtc支持的视频codec类型,在前面的优先级更高
 #以下范例为所有支持的视频codec
 preferredCodecV=H264,H265,AV1,VP9,VP8

ext-codec/AACRtmp.cpp

@@ -17,7 +17,7 @@ using namespace toolkit;
 namespace mediakit {
 
 void AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {
-    CHECK(pkt->size() > 2);
+    CHECK_RET(pkt->size() > 2);
     if (pkt->isConfigFrame()) {
         getTrack()->setExtraData((uint8_t *)pkt->data() + 2, pkt->size() - 2);
         return;

ext-codec/H264Rtmp.cpp

@@ -14,14 +14,6 @@
 using namespace std;
 using namespace toolkit;
 
-#define CHECK_RET(...)                                                         \
-    try {                                                                      \
-        CHECK(__VA_ARGS__);                                                    \
-    } catch (AssertFailedException & ex) {                                     \
-        WarnL << ex.what();                                                    \
-        return;                                                                \
-    }
-
 namespace mediakit {
 
 void H264RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {

ext-codec/H265Rtmp.cpp

@@ -18,14 +18,6 @@
 using namespace std;
 using namespace toolkit;
 
-#define CHECK_RET(...)                                                         \
-    try {                                                                      \
-        CHECK(__VA_ARGS__);                                                    \
-    } catch (AssertFailedException & ex) {                                     \
-        WarnL << ex.what();                                                    \
-        return;                                                                \
-    }
-
 namespace mediakit {
 
 void H265RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {

server/VideoStack.cpp

@@ -1,4 +1,4 @@
-#if defined(ENABLE_X264) && defined(ENABLE_FFMPEG)
+#if defined(ENABLE_VIDEOSTACK) && defined(ENABLE_X264) && defined(ENABLE_FFMPEG)
 #include "VideoStack.h"
 #include "Codec/Transcode.h"
 #include "Common/Device.h"
@@ -62,7 +62,8 @@ void Channel::addParam(const std::weak_ptr<Param>& p)
 void Channel::onFrame(const mediakit::FFmpegFrame::Ptr& frame)
 {
     std::weak_ptr<Channel> weakSelf = shared_from_this();
-    // toolkit::WorkThreadPool::Instance().getFirstPoller()->async([weakSelf, frame]() {
+    _poller = _poller ? _poller : toolkit::WorkThreadPool::Instance().getPoller();
+    _poller->async([weakSelf, frame]() {
         auto self = weakSelf.lock();
         if (!self) {
             return;
@@ -70,7 +71,7 @@ void Channel::onFrame(const mediakit::FFmpegFrame::Ptr& frame)
         self->_tmp = self->_sws->inputFrame(frame);
 
         self->forEachParam([self](const Param::Ptr& p) { self->fillBuffer(p); });
-    // });
+    });
 }
 
 void Channel::forEachParam(const std::function<void(const Param::Ptr&)>& func)
@@ -180,7 +181,7 @@ void StackPlayer::play()
                 self->onFrame(frame);
             });
 
-            videoTrack->addDelegate((std::function<bool(const mediakit::Frame::Ptr&)>)[decoder](const mediakit::Frame::Ptr& frame) {
+            videoTrack->addDelegate([decoder](const mediakit::Frame::Ptr& frame) {
                 return decoder->inputFrame(frame, false, true);
             });
         }
@@ -440,6 +441,7 @@ int VideoStackManager::stopVideoStack(const std::string& id)
     auto it = _stackMap.find(id);
     if (it != _stackMap.end()) {
         _stackMap.erase(it);
+        InfoL << "VideoStack stop: " << id;
         return 0;
     }
     return -1;
@@ -467,8 +469,8 @@ Params VideoStackManager::parseParams(const Json::Value& json,
         float gaph = json["gaph"].asFloat(); //水平间距
 
         //单个间距
-        int gaphPix = static_cast<int>(std::round(width * gaph));
+        int gaphPix = static_cast<int>(round(width * gaph));
-        int gapvPix = static_cast<int>(std::round(height * gapv));
+        int gapvPix = static_cast<int>(round(height * gapv));
 
         // 根据间距计算格子宽高
         int gridWidth = cols > 1 ? (width - gaphPix * (cols - 1)) / cols : width;

server/VideoStack.h

@@ -1,5 +1,5 @@
 #pragma once
-#if defined(ENABLE_X264) && defined(ENABLE_FFMPEG)
+#if defined(ENABLE_VIDEOSTACK) && defined(ENABLE_X264) && defined(ENABLE_FFMPEG)
 #include "Codec/Transcode.h"
 #include "Common/Device.h"
 #include "Player/MediaPlayer.h"
@@ -80,6 +80,7 @@ class Channel : public std::enable_shared_from_this<Channel> {
     std::vector<std::weak_ptr<Param>> _params;
 
     mediakit::FFmpegSws::Ptr _sws;
+    toolkit::EventPoller::Ptr _poller;
 };
 
 class StackPlayer : public std::enable_shared_from_this<StackPlayer> {

server/WebApi.cpp

@@ -59,10 +59,10 @@
 #endif
 
 #if defined(ENABLE_VERSION)
-#include "version.h"
+#include "ZLMVersion.h"
 #endif
 
-#if defined(ENABLE_X264) && defined (ENABLE_FFMPEG)
+#if defined(ENABLE_VIDEOSTACK) && defined(ENABLE_X264) && defined (ENABLE_FFMPEG)
 #include "VideoStack.h"
 #endif
 
@@ -119,7 +119,7 @@ static HttpApi toApi(const function<void(API_ARGS_MAP_ASYNC)> &cb) {
 
         //参数解析成map
         auto args = getAllArgs(parser);
-        cb(sender, headerOut, HttpAllArgs<decltype(args)>(parser, args), val, invoker);
+        cb(sender, headerOut, ArgsMap(parser, args), val, invoker);
     };
 }
 
@@ -147,7 +147,7 @@ static HttpApi toApi(const function<void(API_ARGS_JSON_ASYNC)> &cb) {
         Json::Reader reader;
         reader.parse(parser.content(), args);
 
-        cb(sender, headerOut, HttpAllArgs<decltype(args)>(parser, args), val, invoker);
+        cb(sender, headerOut, ArgsJson(parser, args), val, invoker);
     };
 }
 
@@ -167,7 +167,7 @@ static HttpApi toApi(const function<void(API_ARGS_STRING_ASYNC)> &cb) {
         Json::Value val;
         val["code"] = API::Success;
 
-        cb(sender, headerOut, HttpAllArgs<string>(parser, (string &)parser.content()), val, invoker);
+        cb(sender, headerOut, ArgsString(parser, (string &)parser.content()), val, invoker);
     };
 }
 
@@ -389,6 +389,7 @@ void dumpMediaTuple(const MediaTuple &tuple, Json::Value& item) {
     item[VHOST_KEY] = tuple.vhost;
     item["app"] = tuple.app;
     item["stream"] = tuple.stream;
+    item["params"] = tuple.params;
 }
 
 Value makeMediaSourceJson(MediaSource &media){
@@ -584,8 +585,10 @@ void addStreamProxy(const string &vhost, const string &app, const string &stream
     //添加拉流代理
     auto player = s_player_proxy.make(key, vhost, app, stream, option, retry_count);
 
-    // 先透传参数
+    // 先透传拷贝参数
-    player->mINI::operator=(args);
+    for (auto &pr : args) {
+        (*player)[pr.first] = pr.second;
+    }
 
     //指定RTP over TCP(播放rtsp时有效)
     (*player)[Client::kRtpType] = rtp_type;
@@ -660,13 +663,6 @@ void addStreamPusherProxy(const string &schema,
     pusher->publish(url);
 }
 
-template <typename Type>
-static void getArgsValue(const HttpAllArgs<ApiArgsType> &allArgs, const string &key, Type &value) {
-    auto val = allArgs[key];
-    if (!val.empty()) {
-        value = (Type)val;
-    }
-}
 
 /**
  * 安装api接口
@@ -733,7 +729,7 @@ void installWebApi() {
         CHECK_SECRET();
         auto &ini = mINI::Instance();
         int changed = API::Success;
-        for (auto &pr : allArgs.getArgs()) {
+        for (auto &pr : allArgs.args) {
             if (ini.find(pr.first) == ini.end()) {
 #if 1
                 //没有这个key
@@ -1091,7 +1087,7 @@ void installWebApi() {
         CHECK_ARGS("vhost","app","stream","url");
 
         mINI args;
-        for (auto &pr : allArgs.getArgs()) {
+        for (auto &pr : allArgs.args) {
             args.emplace(pr.first, pr.second);
         }
 
@@ -1188,7 +1184,7 @@ void installWebApi() {
     //测试url http://127.0.0.1/index/api/downloadBin
     api_regist("/index/api/downloadBin",[](API_ARGS_MAP_ASYNC){
         CHECK_SECRET();
-        invoker.responseFile(allArgs.getParser().getHeader(),StrCaseMap(),exePath());
+        invoker.responseFile(allArgs.parser.getHeader(), StrCaseMap(), exePath());
     });
 
 #if defined(ENABLE_RTPPROXY)
@@ -1319,7 +1315,7 @@ void installWebApi() {
         if (!src) {
             throw ApiRetException("can not find the source stream", API::NotFound);
         }
-        auto type = allArgs["type"].as<int>();
+        auto type = allArgs["type"].empty() ? (int)MediaSourceEvent::SendRtpArgs::kRtpPS : allArgs["type"].as<int>();
         if (!allArgs["use_ps"].empty()) {
             // 兼容之前的use_ps参数
             type = allArgs["use_ps"].as<int>();
@@ -1359,7 +1355,7 @@ void installWebApi() {
         if (!src) {
             throw ApiRetException("can not find the source stream", API::NotFound);
         }
-        auto type = allArgs["type"].as<int>();
+        auto type = allArgs["type"].empty() ? (int)MediaSourceEvent::SendRtpArgs::kRtpPS : allArgs["type"].as<int>();
         if (!allArgs["use_ps"].empty()) {
             // 兼容之前的use_ps参数
             type = allArgs["use_ps"].as<int>();
@@ -1568,7 +1564,7 @@ void installWebApi() {
     api_regist("/index/api/deleteRecordDirectory", [](API_ARGS_MAP) {
         CHECK_SECRET();
         CHECK_ARGS("vhost", "app", "stream", "period");
-        auto tuple = MediaTuple{allArgs["vhost"], allArgs["app"], allArgs["stream"]};
+        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 + "/";
@@ -1607,7 +1603,7 @@ void installWebApi() {
     api_regist("/index/api/getMP4RecordFile", [](API_ARGS_MAP){
         CHECK_SECRET();
         CHECK_ARGS("vhost", "app", "stream");
-        auto tuple = MediaTuple{allArgs["vhost"], allArgs["app"], allArgs["stream"]};
+        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"];
 
@@ -1695,7 +1691,7 @@ void installWebApi() {
 
             //截图存在，且未过期，那么返回之
             res_old_snap = true;
-            responseSnap(path, allArgs.getParser().getHeader(), invoker);
+            responseSnap(path, allArgs.parser.getHeader(), invoker);
             //中断遍历
             return false;
         });
@@ -1726,7 +1722,7 @@ void installWebApi() {
                 File::delete_file(new_snap);
                 rename(new_snap_tmp.data(), new_snap.data());
             }
-            responseSnap(new_snap, allArgs.getParser().getHeader(), invoker, err_msg);
+            responseSnap(new_snap, allArgs.parser.getHeader(), invoker, err_msg);
         });
     });
 
@@ -1741,7 +1737,7 @@ void installWebApi() {
 #ifdef ENABLE_WEBRTC
     class WebRtcArgsImp : public WebRtcArgs {
     public:
-        WebRtcArgsImp(const HttpAllArgs<string> &args, std::string session_id)
+        WebRtcArgsImp(const ArgsString &args, std::string session_id)
             : _args(args)
             , _session_id(std::move(session_id)) {}
         ~WebRtcArgsImp() override = default;
@@ -1759,40 +1755,26 @@ void installWebApi() {
             CHECK_ARGS("app", "stream");
 
             return StrPrinter << "rtc://" << _args["Host"] << "/" << _args["app"] << "/"
-                              << _args["stream"] << "?" << _args.getParser().params() + "&session=" + _session_id;
+                              << _args["stream"] << "?" << _args.parser.params() + "&session=" + _session_id;
         }
 
     private:
-        HttpAllArgs<string> _args;
+        ArgsString _args;
         std::string _session_id;
     };
 
     api_regist("/index/api/webrtc",[](API_ARGS_STRING_ASYNC){
         CHECK_ARGS("type");
         auto type = allArgs["type"];
-        auto offer = allArgs.getArgs();
+        auto offer = allArgs.args;
         CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
-        std::string host = allArgs.getParser()["Host"];
-        std::string localIp = host.substr(0, host.find(':'));
-
-        auto isVaildIP = [](std::string ip)-> bool {
-            int a,b,c,d;
-            return sscanf(ip.c_str(),"%d.%d.%d.%d", &a, &b, &c, &d) == 4;
-        };
-        if (!isVaildIP(localIp) || localIp=="127.0.0.1") {
-            localIp = "";
-        }
 
+        auto &session = static_cast<Session&>(sender);
         auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
-        WebRtcPluginManager::Instance().getAnswerSdp(static_cast<Session&>(sender), type, *args, [invoker, val, offer, headerOut, localIp](const WebRtcInterface &exchanger) mutable {
+        WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, val, offer, headerOut](const WebRtcInterface &exchanger) mutable {
-            //设置返回类型
+            auto &handler = const_cast<WebRtcInterface &>(exchanger);
-            headerOut["Content-Type"] = HttpFileManager::getContentType(".json");
-            //设置跨域
-            headerOut["Access-Control-Allow-Origin"] = "*";
-
             try {
-                setLocalIp(exchanger,localIp);
+                val["sdp"] = handler.getAnswerSdp(offer);
-                val["sdp"] = exchangeSdp(exchanger, offer);
                 val["id"] = exchanger.getIdentifier();
                 val["type"] = "answer";
                 invoker(200, headerOut, val.toStyledString());
@@ -1806,21 +1788,19 @@ void installWebApi() {
 
     static constexpr char delete_webrtc_url [] = "/index/api/delete_webrtc";
     static auto whip_whep_func = [](const char *type, API_ARGS_STRING_ASYNC) {
-        auto offer = allArgs.getArgs();
+        auto offer = allArgs.args;
         CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
 
         auto &session = static_cast<Session&>(sender);
-        auto location = std::string("http") + (session.overSsl() ? "s" : "") + "://" + allArgs["host"] + delete_webrtc_url;
+        auto location = std::string(session.overSsl() ? "https://" : "http://") + allArgs["host"] + delete_webrtc_url;
         auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
-        WebRtcPluginManager::Instance().getAnswerSdp(session, type, *args,
+        WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, offer, headerOut, location](const WebRtcInterface &exchanger) mutable {
-                                                     [invoker, offer, headerOut, location](const WebRtcInterface &exchanger) mutable {
+            auto &handler = const_cast<WebRtcInterface &>(exchanger);
-                // 设置跨域
-                headerOut["Access-Control-Allow-Origin"] = "*";
             try {
                 // 设置返回类型
                 headerOut["Content-Type"] = "application/sdp";
                 headerOut["Location"] = location + "?id=" + exchanger.getIdentifier() + "&token=" + exchanger.deleteRandStr();
-                    invoker(201, headerOut, exchangeSdp(exchanger, offer));
+                invoker(201, headerOut, handler.getAnswerSdp(offer));
             } catch (std::exception &ex) {
                 headerOut["Content-Type"] = "text/plain";
                 invoker(406, headerOut, ex.what());
@@ -1833,7 +1813,7 @@ void installWebApi() {
 
     api_regist(delete_webrtc_url, [](API_ARGS_MAP_ASYNC) {
         CHECK_ARGS("id", "token");
-        CHECK(allArgs.getParser().method() == "DELETE", "http method is not DELETE: " + allArgs.getParser().method());
+        CHECK(allArgs.parser.method() == "DELETE", "http method is not DELETE: " + allArgs.parser.method());
         auto obj = WebRtcTransportManager::Instance().getItem(allArgs["id"]);
         if (!obj) {
             invoker(404, headerOut, "id not found");
@@ -1919,44 +1899,39 @@ void installWebApi() {
                 if (!save_name.empty()) {
                     res_header.emplace("Content-Disposition", "attachment;filename=\"" + save_name + "\"");
                 }
-                invoker.responseFile(allArgs.getParser().getHeader(), res_header, allArgs["file_path"]);
+                invoker.responseFile(allArgs.parser.getHeader(), res_header, allArgs["file_path"]);
             }
         };
 
-        bool flag = NOTICE_EMIT(BroadcastHttpAccessArgs, Broadcast::kBroadcastHttpAccess, allArgs.getParser(), file_path, false, file_invoker, sender);
+        bool flag = NOTICE_EMIT(BroadcastHttpAccessArgs, Broadcast::kBroadcastHttpAccess, allArgs.parser, file_path, false, file_invoker, sender);
         if (!flag) {
             // 文件下载鉴权事件无人监听，不允许下载
             invoker(401, StrCaseMap {}, "None http access event listener");
         }
     });
 
-#if defined(ENABLE_X264) && defined(ENABLE_FFMPEG)
+#if defined(ENABLE_VIDEOSTACK) && defined(ENABLE_X264) && defined(ENABLE_FFMPEG)
     VideoStackManager::Instance().loadBgImg("novideo.yuv");
     NoticeCenter::Instance().addListener(nullptr, Broadcast::kBroadcastStreamNoneReader, [](BroadcastStreamNoneReaderArgs) {
         auto id = sender.getMediaTuple().stream;
         VideoStackManager::Instance().stopVideoStack(id);
-        InfoL << "VideoStack: " << id <<" stop";
     });
 
     api_regist("/index/api/stack/start", [](API_ARGS_JSON_ASYNC) {
         CHECK_SECRET();
-        auto ret = VideoStackManager::Instance().startVideoStack(allArgs.getArgs());
+        auto ret = VideoStackManager::Instance().startVideoStack(allArgs.args);
-        if (!ret) {
+        val["code"] = ret;
-            invoker(200, headerOut, "success");
+        val["msg"] = ret ? "failed" : "success";
-        } else {
+        invoker(200, headerOut, val.toStyledString());
-            invoker(200, headerOut, "failed");
-        }
     });
 
     api_regist("/index/api/stack/stop", [](API_ARGS_MAP_ASYNC) {
         CHECK_SECRET();
         CHECK_ARGS("id");
         auto ret = VideoStackManager::Instance().stopVideoStack(allArgs["id"]);
-        if (!ret) {
+        val["code"] = ret;
-            invoker(200, headerOut, "success");
+        val["msg"] = ret ? "failed" : "success";
-        } else {
+        invoker(200, headerOut, val.toStyledString());
-            invoker(200, headerOut, "failed");
-        }
     });
 #endif
 }

server/WebApi.h

@@ -115,72 +115,41 @@ std::string getValue(const mediakit::Parser &parser, Args &args, const First &fi
 
 template<typename Args>
 class HttpAllArgs {
+    mediakit::Parser* _parser = nullptr;
+    Args* _args = nullptr;
 public:
-    HttpAllArgs(const mediakit::Parser &parser, Args &args) {
+    const mediakit::Parser& parser;
-        _get_args = [&args]() {
+    Args& args;
-            return (void *) &args;
-        };
-        _get_parser = [&parser]() -> const mediakit::Parser & {
-            return parser;
-        };
-        _get_value = [](HttpAllArgs &that, const std::string &key) {
-            return getValue(that.getParser(), that.getArgs(), key);
-        };
-        _clone = [&](HttpAllArgs &that) {
-            that._get_args = [args]() {
-                return (void *) &args;
-            };
-            that._get_parser = [parser]() -> const mediakit::Parser & {
-                return parser;
-            };
-            that._get_value = [](HttpAllArgs &that, const std::string &key) {
-                return getValue(that.getParser(), that.getArgs(), key);
-            };
-            that._cache_able = true;
-        };
-    }
 
-    HttpAllArgs(const HttpAllArgs &that) {
+    HttpAllArgs(const mediakit::Parser &p, Args &a): parser(p), args(a) {}
-        if (that._cache_able) {
+
-            _get_args = that._get_args;
+    HttpAllArgs(const HttpAllArgs &that): _parser(new mediakit::Parser(that.parser)),
-            _get_parser = that._get_parser;
+                                          _args(new Args(that.args)),
-            _get_value = that._get_value;
+                                          parser(*_parser), args(*_args) {}
-            _cache_able = true;
+    ~HttpAllArgs() {
-        } else {
+        if (_parser) {
-            that._clone(*this);
+            delete _parser;
+        }
+        if (_args) {
+            delete _args;
         }
     }
 
     template<typename Key>
     toolkit::variant operator[](const Key &key) const {
-        return (toolkit::variant)_get_value(*(HttpAllArgs*)this, key);
+        return (toolkit::variant)getValue(parser, args, key);
     }
-
-    const mediakit::Parser &getParser() const {
-        return _get_parser();
-    }
-
-    Args &getArgs() {
-        return *((Args *) _get_args());
-    }
-
-    const Args &getArgs() const {
-        return *((Args *) _get_args());
-    }
-
-private:
-    bool _cache_able = false;
-    std::function<void *() > _get_args;
-    std::function<const mediakit::Parser &() > _get_parser;
-    std::function<std::string(HttpAllArgs &that, const std::string &key)> _get_value;
-    std::function<void(HttpAllArgs &that) > _clone;
 };
 
-#define API_ARGS_MAP toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<ApiArgsType> &allArgs, Json::Value &val
+using ArgsMap = HttpAllArgs<ApiArgsType>;
+using ArgsJson = HttpAllArgs<Json::Value>;
+using ArgsString = HttpAllArgs<std::string>;
+
+#define API_ARGS_MAP toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsMap &allArgs, Json::Value &val
 #define API_ARGS_MAP_ASYNC API_ARGS_MAP, const mediakit::HttpSession::HttpResponseInvoker &invoker
-#define API_ARGS_JSON toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<Json::Value> &allArgs, Json::Value &val
+#define API_ARGS_JSON toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsJson &allArgs, Json::Value &val
 #define API_ARGS_JSON_ASYNC API_ARGS_JSON, const mediakit::HttpSession::HttpResponseInvoker &invoker
-#define API_ARGS_STRING toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<std::string> &allArgs, Json::Value &val
+#define API_ARGS_STRING toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsString &allArgs, Json::Value &val
 #define API_ARGS_STRING_ASYNC API_ARGS_STRING, const mediakit::HttpSession::HttpResponseInvoker &invoker
 #define API_ARGS_VALUE sender, headerOut, allArgs, val
 
@@ -234,8 +203,6 @@ 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, int only_track, 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
 
 Json::Value makeMediaSourceJson(mediakit::MediaSource &media);

server/WebHook.cpp

@@ -225,7 +225,7 @@ static ArgsType make_json(const MediaInfo &args) {
     ArgsType body;
     body["schema"] = args.schema;
     dumpMediaTuple(args, body);
-    body["params"] = args.param_strs;
+    body["params"] = args.params;
     return body;
 }
 
@@ -286,7 +286,7 @@ static string getPullUrl(const string &origin_fmt, const MediaInfo &info) {
         return "";
     }
     // 告知源站这是来自边沿站的拉流请求，如果未找到流请立即返回拉流失败
-    return string(url) + '?' + kEdgeServerParam + '&' + VHOST_KEY + '=' + info.vhost + '&' + info.param_strs;
+    return string(url) + '?' + kEdgeServerParam + '&' + VHOST_KEY + '=' + info.vhost + '&' + info.params;
 }
 
 static void pullStreamFromOrigin(const vector<string> &urls, size_t index, size_t failed_cnt, const MediaInfo &args, const function<void()> &closePlayer) {
@@ -498,7 +498,7 @@ void installWebHook() {
             return;
         }
 
-        if (start_with(args.param_strs, kEdgeServerParam)) {
+        if (start_with(args.params, kEdgeServerParam)) {
             // 源站收到来自边沿站的溯源请求，流不存在时立即返回拉流失败
             closePlayer();
             return;

server/main.cpp

@@ -38,7 +38,7 @@
 #endif
 
 #if defined(ENABLE_VERSION)
-#include "version.h"
+#include "ZLMVersion.h"
 #endif
 
 #if !defined(_WIN32)
@@ -258,6 +258,15 @@ int start_main(int argc,char *argv[]) {
         //加载配置文件，如果配置文件不存在就创建一个
         loadIniConfig(g_ini_file.data());
 
+        auto &secret = mINI::Instance()[API::kSecret];
+        if (secret == "035c73f7-bb6b-4889-a715-d9eb2d1925cc" || secret.empty()) {
+            // 使用默认secret被禁止启动
+            secret = makeRandStr(32, true);
+            mINI::Instance().dumpFile(g_ini_file);
+            WarnL << "The " << API::kSecret << " is invalid, modified it to: " << secret
+                  << ", saved config file: " << g_ini_file;
+        }
+
         if (!File::is_dir(ssl_file)) {
             // 不是文件夹，加载证书，证书包含公钥和私钥
             SSL_Initor::Instance().loadCertificate(ssl_file.data());
@@ -352,14 +361,6 @@ int start_main(int argc,char *argv[]) {
         InfoL << "已启动http hook 接口";
 
         try {
-            auto &secret = mINI::Instance()[API::kSecret];
-            if (secret == "035c73f7-bb6b-4889-a715-d9eb2d1925cc" || secret.empty()) {
-                // 使用默认secret被禁止启动
-                secret = makeRandStr(32, true);
-                mINI::Instance().dumpFile(g_ini_file);
-                WarnL << "The " << API::kSecret << " is invalid, modified it to: " << secret
-                      << ", saved config file: " << g_ini_file;
-            }
             //rtsp服务器，端口默认554
             if (rtspPort) { rtspSrv->start<RtspSession>(rtspPort); }
             //rtsps服务器，端口默认322

src/Common/MediaSource.cpp

@@ -113,7 +113,7 @@ ProtocolOption::ProtocolOption() {
 //////////////////////////////////////////////////////////////////////////////////////////////////////////////
 
 struct MediaSourceNull : public MediaSource {
-    MediaSourceNull() : MediaSource("schema", MediaTuple{"vhost", "app", "stream"}) {};
+    MediaSourceNull() : MediaSource("schema", MediaTuple{"vhost", "app", "stream", ""}) {};
     int readerCount() override { return 0; }
 };
 
@@ -583,7 +583,7 @@ void MediaInfo::parse(const std::string &url_in){
     auto url = url_in;
     auto pos = url.find("?");
     if (pos != string::npos) {
-        param_strs = url.substr(pos + 1);
+        params = url.substr(pos + 1);
         url.erase(pos);
     }
 
@@ -616,9 +616,10 @@ void MediaInfo::parse(const std::string &url_in){
         stream = stream_id;
     }
 
-    auto params = Parser::parseArgs(param_strs);
+    auto kv = Parser::parseArgs(params);
-    if (params.find(VHOST_KEY) != params.end()) {
+    auto it = kv.find(VHOST_KEY);
-        vhost = params[VHOST_KEY];
+    if (it != kv.end()) {
+        vhost = it->second;
     }
 
     GET_CONFIG(bool, enableVhost, General::kEnableVhost);

src/Common/MediaSource.h

@@ -136,6 +136,15 @@ private:
     toolkit::Timer::Ptr _async_close_timer;
 };
 
+
+template <typename MAP, typename KEY, typename TYPE>
+static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
+    auto val = ((MAP &)allArgs)[key];
+    if (!val.empty()) {
+        value = (TYPE)val;
+    }
+}
+
 class ProtocolOption {
 public:
     ProtocolOption();
@@ -243,15 +252,6 @@ public:
         GET_OPT_VALUE(stream_replace);
         GET_OPT_VALUE(max_track);
     }
-
-private:
-    template <typename MAP, typename KEY, typename TYPE>
-    static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
-        auto val = ((MAP &)allArgs)[key];
-        if (!val.empty()) {
-            value = (TYPE)val;
-        }
-    }
 };
 
 //该对象用于拦截感兴趣的MediaSourceEvent事件
@@ -299,7 +299,6 @@ public:
     std::string full_url;
     std::string schema;
     std::string host;
-    std::string param_strs;
 };
 
 bool equalMediaTuple(const MediaTuple& a, const MediaTuple& b);

src/Common/Parser.cpp

@@ -294,8 +294,8 @@ void RtspUrl::setup(bool is_ssl, const string &url, const string &user, const st
     splitUrl(ip, ip, port);
 
     _url = std::move(url);
-    _user = strCoding::UrlDecode(std::move(user));
+    _user = strCoding::UrlDecodeUserOrPass(user);
-    _passwd = strCoding::UrlDecode(std::move(passwd));
+    _passwd = strCoding::UrlDecodeUserOrPass(passwd);
     _host = std::move(ip);
     _port = port;
     _is_ssl = is_ssl;

src/Common/Parser.h

@@ -30,7 +30,7 @@ struct StrCaseCompare {
 
 class StrCaseMap : public std::multimap<std::string, std::string, StrCaseCompare> {
 public:
-    using Super = multimap<std::string, std::string, StrCaseCompare>;
+    using Super = std::multimap<std::string, std::string, StrCaseCompare>;
 
     std::string &operator[](const std::string &k) {
         auto it = find(k);

src/Common/macros.cpp

@@ -14,7 +14,7 @@
 using namespace toolkit;
 
 #if defined(ENABLE_VERSION)
-#include "version.h"
+#include "ZLMVersion.h"
 #endif
 
 extern "C" {

src/Common/macros.h

@@ -36,6 +36,16 @@
 #define CHECK(exp, ...) ::mediakit::Assert_ThrowCpp(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__, ##__VA_ARGS__)
 #endif // CHECK
 
+#ifndef CHECK_RET
+#define CHECK_RET(...)                                                         \
+    try {                                                                      \
+        CHECK(__VA_ARGS__);                                                    \
+    } catch (AssertFailedException & ex) {                                     \
+        WarnL << ex.what();                                                    \
+        return;                                                                \
+    }
+#endif
+
 #ifndef MAX
 #define MAX(a, b) ((a) > (b) ? (a) : (b))
 #endif // MAX

src/Common/strCoding.cpp

@@ -52,25 +52,7 @@ char HexStrToBin(const char *str) {
     }
     return (high << 4) | low;
 }
-
+static string UrlEncodeCommon(const string &str,const char* dont_escape){
-string strCoding::UrlEncode(const string &str) {
-    string out;
-    size_t len = str.size();
-    for (size_t i = 0; i < len; ++i) {
-        char ch = str[i];
-        if (isalnum((uint8_t) ch)) {
-            out.push_back(ch);
-        } else {
-            char buf[4];
-            sprintf(buf, "%%%X%X", (uint8_t) ch >> 4, (uint8_t) ch & 0x0F);
-            out.append(buf);
-        }
-    }
-    return out;
-}
-
-string strCoding::UrlEncodePath(const string &str) {
-    const char *dont_escape = "!#&'*+:=?@/._-$,;~()";
     string out;
     size_t len = str.size();
     for (size_t i = 0; i < len; ++i) {
@@ -85,52 +67,7 @@ string strCoding::UrlEncodePath(const string &str) {
     }
     return out;
 }
-
+static string UrlDecodeCommon(const string &str,const char* dont_unescape){
-string strCoding::UrlEncodeComponent(const string &str) {
-    const char *dont_escape = "!'()*-._~";
-    string out;
-    size_t len = str.size();
-    for (size_t i = 0; i < len; ++i) {
-        char ch = str[i];
-        if (isalnum((uint8_t) ch) || strchr(dont_escape, (uint8_t) ch) != NULL) {
-            out.push_back(ch);
-        } else {
-            char buf[4];
-            snprintf(buf, 4, "%%%X%X", (uint8_t) ch >> 4, (uint8_t) ch & 0x0F);
-            out.append(buf);
-        }
-    }
-    return out;
-}
-
-string strCoding::UrlDecode(const string &str) {
-    string output;
-    size_t i = 0, len = str.length();
-    while (i < len) {
-        if (str[i] == '%') {
-            if (i + 3 > len) {
-                // %后面必须还有两个字节才会反转义
-                output.append(str, i, len - i);
-                break;
-            }
-            char ch = HexStrToBin(&(str[i + 1]));
-            if (ch == -1) {
-                // %后面两个字节不是16进制字符串，转义失败；直接拼接3个原始字符
-                output.append(str, i, 3);
-            } else {
-                output += ch;
-            }
-            i += 3;
-        } else {
-            output += str[i];
-            ++i;
-        }
-    }
-    return output;
-}
-
-string strCoding::UrlDecodePath(const string &str) {
-    const char *dont_unescape = "#$&+,/:;=?@";
     string output;
     size_t i = 0, len = str.length();
     while (i < len) {
@@ -156,6 +93,36 @@ string strCoding::UrlDecodePath(const string &str) {
     return output;
 }
 
+string strCoding::UrlEncodePath(const string &str) {
+    const char *dont_escape = "!#&'*+:=?@/._-$,;~()";
+    return UrlEncodeCommon(str,dont_escape);
+}
+
+string strCoding::UrlEncodeComponent(const string &str) {
+    const char *dont_escape = "!'()*-._~";
+    return UrlEncodeCommon(str,dont_escape);
+}
+
+std::string strCoding::UrlEncodeUserOrPass(const std::string &str) {
+    // from rfc https://datatracker.ietf.org/doc/html/rfc3986
+    // §2.3 Unreserved characters (mark)
+    //'-', '_', '.', '~'  
+    //  §2.2 Reserved characters (reserved)
+    // '$', '&', '+', ',', '/', ':', ';', '=', '?', '@', 
+    // §3.2.1
+    // The RFC allows ';', ':', '&', '=', '+', '$', and ',' in
+    // userinfo, so we must escape only '@', '/', and '?'.
+    // The parsing of userinfo treats ':' as special so we must escape
+    // that too.
+    const char *dont_escape = "$&+,;=-._~";
+    return UrlEncodeCommon(str,dont_escape);
+}
+
+string strCoding::UrlDecodePath(const string &str) {
+    const char *dont_unescape = "#$&+,/:;=?@";
+    return UrlDecodeCommon(str,dont_unescape);
+}
+
 std::string strCoding::UrlDecodeComponent(const std::string &str) {
     string output;
     size_t i = 0, len = str.length();
@@ -185,27 +152,11 @@ std::string strCoding::UrlDecodeComponent(const std::string &str) {
     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
 
+std::string strCoding::UrlDecodeUserOrPass(const std::string &str) {
+    const char *dont_unescape = "";
+    return UrlDecodeCommon(str,dont_unescape);
+}
 ///////////////////////////////windows专用///////////////////////////////////
 #if defined(_WIN32)
 void UnicodeToGB2312(char* pOut, wchar_t uData)

src/Common/strCoding.h

@@ -18,12 +18,12 @@ namespace mediakit {
 
 class strCoding {
 public:
-    [[deprecated]] static std::string UrlEncode(const std::string &str); //url utf8编码, deprecated
     static std::string UrlEncodePath(const std::string &str); //url路径 utf8编码
     static std::string UrlEncodeComponent(const std::string &str); // url参数 utf8编码
-    [[deprecated]] static std::string UrlDecode(const std::string &str); //url utf8解码, deprecated
     static std::string UrlDecodePath(const std::string &str); //url路径 utf8解码
     static std::string UrlDecodeComponent(const std::string &str); // url参数 utf8解码
+    static std::string UrlEncodeUserOrPass(const std::string &str); // url中用户名与密码编码
+    static std::string UrlDecodeUserOrPass(const std::string &str); // url中用户名与密码解码
 #if defined(_WIN32)
     static std::string UTF8ToGB2312(const std::string &str);//utf_8转为gb2312
     static std::string GB2312ToUTF8(const std::string &str); //gb2312 转utf_8

src/Http/HttpRequestSplitter.cpp

@@ -65,18 +65,18 @@ void HttpRequestSplitter::input(const char *data,size_t len) {
         _content_len = onRecvHeader(header_ptr, header_size);
     }
 
-    if(_remain_data_size <= 0){
-        //没有剩余数据，清空缓存
-        _remain_data.clear();
-        return;
-    }
-
     /*
      * 恢复末尾字节
      * 移动到这来，目的是防止HttpRequestSplitter::reset()导致内存失效
      */
     tail_ref = tail_tmp;
 
+    if(_remain_data_size <= 0){
+        //没有剩余数据，清空缓存
+        _remain_data.clear();
+        return;
+    }
+
     if(_content_len == 0){
         //尚未找到http头，缓存定位到剩余数据部分
         _remain_data.assign(ptr,_remain_data_size);

src/Http/HttpSession.cpp

@@ -683,18 +683,6 @@ void HttpSession::sendResponse(int code,
     AsyncSender::onSocketFlushed(data);
 }
 
-string HttpSession::urlDecode(const string &str) {
-    auto ret = strCoding::UrlDecode(str);
-#ifdef _WIN32
-    GET_CONFIG(string, charSet, Http::kCharSet);
-    bool isGb2312 = !strcasecmp(charSet.data(), "gb2312");
-    if (isGb2312) {
-        ret = strCoding::UTF8ToGB2312(ret);
-    }
-#endif // _WIN32
-    return ret;
-}
-
 string HttpSession::urlDecodePath(const string &str) {
     auto ret = strCoding::UrlDecodePath(str);
 #ifdef _WIN32

src/Http/HttpSession.h

@@ -44,7 +44,6 @@ public:
     void onRecv(const toolkit::Buffer::Ptr &) override;
     void onError(const toolkit::SockException &err) override;
     void onManager() override;
-    [[deprecated]] static std::string urlDecode(const std::string &str);
     static std::string urlDecodePath(const std::string &str);
     static std::string urlDecodeComponent(const std::string &str);
     void setTimeoutSec(size_t second);

src/Record/HlsMakerImp.cpp

@@ -195,10 +195,8 @@ std::shared_ptr<FILE> HlsMakerImp::makeFile(const string &file, bool setbuf) {
     return ret;
 }
 
-void HlsMakerImp::setMediaSource(const string &vhost, const string &app, const string &stream_id) {
+void HlsMakerImp::setMediaSource(const MediaTuple& tuple) {
-    _info.app = app;
+    static_cast<MediaTuple &>(_info) = tuple;
-    _info.stream = stream_id;
-    _info.vhost = vhost;
     _media_src = std::make_shared<HlsMediaSource>(isFmp4() ? HLS_FMP4_SCHEMA : HLS_SCHEMA, _info);
 }
 

src/Record/HlsMakerImp.h

@@ -27,11 +27,8 @@ public:
 
     /**
      * 设置媒体信息
-     * @param vhost 虚拟主机
-     * @param app 应用名
-     * @param stream_id 流id
      */
-    void setMediaSource(const std::string &vhost, const std::string &app, const std::string &stream_id);
+    void setMediaSource(const MediaTuple& tuple);
 
     /**
      * 获取MediaSource

src/Record/HlsRecorder.h

@@ -34,7 +34,7 @@ public:
     }
 
     void setMediaSource(const MediaTuple& tuple) {
-        _hls->setMediaSource(tuple.vhost, tuple.app, tuple.stream);
+        _hls->setMediaSource(tuple);
     }
 
     void setListener(const std::weak_ptr<MediaSourceEvent> &listener) {

src/Record/MP4Reader.cpp

@@ -38,7 +38,7 @@ MP4Reader::MP4Reader(const std::string &vhost, const std::string &app, const std
 
 void MP4Reader::setup(const std::string &vhost, const std::string &app, const std::string &stream_id, const std::string &file_path, const ProtocolOption &option, toolkit::EventPoller::Ptr poller) {
     //读写文件建议放在后台线程
-    auto tuple =  MediaTuple{vhost, app, stream_id};
+    auto tuple =  MediaTuple{vhost, app, stream_id, ""};
     _poller = poller ? std::move(poller) : WorkThreadPool::Instance().getPoller();
     _file_path = file_path;
     if (_file_path.empty()) {

src/Record/MP4Recorder.cpp

@@ -22,12 +22,10 @@ using namespace toolkit;
 
 namespace mediakit {
 
-MP4Recorder::MP4Recorder(const string &path, const string &vhost, const string &app, const string &stream_id, size_t max_second) {
+MP4Recorder::MP4Recorder(const MediaTuple &tuple, const string &path, size_t max_second) {
     _folder_path = path;
     /////record 业务逻辑//////
-    _info.app = app;
+    static_cast<MediaTuple &>(_info) = tuple;
-    _info.stream = stream_id;
-    _info.vhost = vhost;
     _info.folder = path;
     GET_CONFIG(uint32_t, s_max_second, Protocol::kMP4MaxSecond);
     _max_second = max_second ? max_second : s_max_second;
@@ -120,7 +118,7 @@ bool MP4Recorder::inputFrame(const Frame::Ptr &frame) {
             //b帧情况下dts时间戳可能回退
             _last_dts = MAX(frame->dts(), _last_dts);
         }
-        auto duration = 5; // 默认至少一帧5ms
+        auto duration = 5u; // 默认至少一帧5ms
         if (frame->dts() > 0 && frame->dts() > _last_dts) {
             duration = MAX(duration, frame->dts() - _last_dts);
         }

src/Record/MP4Recorder.h

@@ -26,7 +26,7 @@ class MP4Recorder final : public MediaSinkInterface {
 public:
     using Ptr = std::shared_ptr<MP4Recorder>;
 
-    MP4Recorder(const std::string &path, const std::string &vhost, const std::string &app, const std::string &stream_id, size_t max_second);
+    MP4Recorder(const MediaTuple &tuple, const std::string &path, size_t max_second);
     ~MP4Recorder() override;
 
     /**

src/Record/Recorder.cpp

@@ -68,8 +68,7 @@ string Recorder::getRecordPath(Recorder::type type, const MediaTuple& tuple, con
             }
             return File::absolutePath(m3u8FilePath, hlsPath);
         }
-        default:
+        default: return "";
-            return "";
     }
 }
 
@@ -85,13 +84,12 @@ std::shared_ptr<MediaSinkInterface> Recorder::createRecorder(type type, const Me
 #else
             throw std::invalid_argument("hls相关功能未打开，请开启ENABLE_HLS宏后编译再测试");
 #endif
-
         }
 
         case Recorder::type_mp4: {
 #if defined(ENABLE_MP4)
             auto path = Recorder::getRecordPath(type, tuple, option.mp4_save_path);
-            return std::make_shared<MP4Recorder>(path, tuple.vhost, tuple.app, tuple.stream, option.mp4_max_second);
+            return std::make_shared<MP4Recorder>(tuple, path, option.mp4_max_second);
 #else
             throw std::invalid_argument("mp4相关功能未打开，请开启ENABLE_MP4宏后编译再测试");
 #endif

src/Record/Recorder.h

@@ -22,6 +22,7 @@ struct MediaTuple {
     std::string vhost;
     std::string app;
     std::string stream;
+    std::string params;
     std::string shortUrl() const {
         return vhost + '/' + app + '/' + stream;
     }

src/Rtmp/RtmpMediaSourceMuxer.h

@@ -29,7 +29,13 @@ public:
         getRtmpRing()->setDelegate(_media_src);
     }
 
-    ~RtmpMediaSourceMuxer() override { RtmpMuxer::flush(); }
+    ~RtmpMediaSourceMuxer() override {
+        try {
+            RtmpMuxer::flush();
+        } catch (std::exception &ex) {
+            WarnL << ex.what();
+        }
+    }
 
     void setListener(const std::weak_ptr<MediaSourceEvent> &listener){
         setDelegate(listener);

src/Rtmp/RtmpProtocol.cpp

@@ -165,14 +165,7 @@ void RtmpProtocol::sendResponse(int type, const string &str) {
 
 void RtmpProtocol::sendInvoke(const string &cmd, const AMFValue &val) {
     AMFEncoder enc;
-    if (val.type() == AMFType::AMF_OBJECT || val.type() == AMFType::AMF_NULL)
-    {
     enc << cmd << ++_send_req_id << val;
-    }
-    else
-    {
-        enc << cmd << ++_send_req_id << AMFValue() << val;
-    }
     sendRequest(MSG_CMD, enc.data());
 }
 
@@ -632,6 +625,7 @@ const char* RtmpProtocol::handle_rtmp(const char *data, size_t len) {
             case 4:
                 chunk_data.ts_field = load_be24(header->time_stamp);
         }
+
         auto time_stamp = chunk_data.ts_field;
         if (chunk_data.ts_field == 0xFFFFFF) {
             if (len < header_len + offset + 4) {

src/Rtmp/RtmpProtocol.h

@@ -11,7 +11,6 @@
 #ifndef SRC_RTMP_RTMPPROTOCOL_H_
 #define SRC_RTMP_RTMPPROTOCOL_H_
 
-#include <cstdint>
 #include <memory>
 #include <string>
 #include <functional>

src/Rtmp/RtmpPusher.cpp

@@ -163,18 +163,6 @@ void RtmpPusher::send_connect() {
 }
 
 void RtmpPusher::send_createStream() {
-    // Workaround : 兼容较旧的 FMS3.0
-    {
-        {
-            AMFValue obj(_stream_id);
-            sendInvoke("releaseStream", obj);
-        }
-        {
-            AMFValue obj(_stream_id);
-            sendInvoke("FCPublish", obj);
-        }
-    }
-    {
     AMFValue obj(AMF_NULL);
     sendInvoke("createStream", obj);
     addOnResultCB([this](AMFDecoder &dec) {
@@ -185,8 +173,6 @@ void RtmpPusher::send_createStream() {
     });
 }
 
-}
-
 #define RTMP_STREAM_LIVE    "live"
 void RtmpPusher::send_publish() {
     AMFEncoder enc;

srt/SrtTransportImp.cpp

@@ -63,8 +63,8 @@ void SrtTransportImp::onHandShakeFinished(std::string &streamid, struct sockaddr
         return;
     }
 
-    auto params = Parser::parseArgs(_media_info.param_strs);
+    auto kv = Parser::parseArgs(_media_info.params);
-    if (params["m"] == "publish") {
+    if (kv["m"] == "publish") {
         _is_pusher = true;
         _decoder = DecoderImp::createDecoder(DecoderImp::decoder_ts, this);
         emitOnPublish();
@@ -98,10 +98,10 @@ bool SrtTransportImp::parseStreamid(std::string &streamid) {
             app = tmps[0];
             stream_name = tmps[1];
         } else {
-            if (_media_info.param_strs.empty()) {
+            if (_media_info.params.empty()) {
-                _media_info.param_strs = it.first + "=" + it.second;
+                _media_info.params = it.first + "=" + it.second;
             } else {
-                _media_info.param_strs += "&" + it.first + "=" + it.second;
+                _media_info.params += "&" + it.first + "=" + it.second;
             }
         }
     }
@@ -118,7 +118,7 @@ bool SrtTransportImp::parseStreamid(std::string &streamid) {
     _media_info.app = app;
     _media_info.stream = stream_name;
 
-    TraceL << " mediainfo=" << _media_info.shortUrl() << " params=" << _media_info.param_strs;
+    TraceL << " mediainfo=" << _media_info.shortUrl() << " params=" << _media_info.params;
 
     return true;
 }

tests/test_flv.cpp

@@ -27,7 +27,7 @@ using namespace mediakit;
 class FlvSplitterImp : public FlvSplitter {
 public:
     FlvSplitterImp() {
-        _src = std::make_shared<RtmpMediaSourceImp>(MediaTuple{DEFAULT_VHOST, "live", "test"});
+        _src = std::make_shared<RtmpMediaSourceImp>(MediaTuple{DEFAULT_VHOST, "live", "test", ""});
     }
     ~FlvSplitterImp()  override = default;
 

tests/test_server.cpp

@@ -90,7 +90,7 @@ void initEventListener() {
     static onceToken s_token([]() {
         //监听kBroadcastOnGetRtspRealm事件决定rtsp链接是否需要鉴权(传统的rtsp鉴权方案)才能访问
         NoticeCenter::Instance().addListener(nullptr, Broadcast::kBroadcastOnGetRtspRealm, [](BroadcastOnGetRtspRealmArgs) {
-             DebugL << "RTSP是否需要鉴权事件：" << args.getUrl() << " " << args.param_strs;
+             DebugL << "RTSP是否需要鉴权事件：" << args.getUrl() << " " << args.params;
              if (string("1") == args.stream) {
                  // live/1需要认证
                  //该流需要认证，并且设置realm
@@ -104,7 +104,7 @@ void initEventListener() {
 
         //监听kBroadcastOnRtspAuth事件返回正确的rtsp鉴权用户密码
         NoticeCenter::Instance().addListener(nullptr, Broadcast::kBroadcastOnRtspAuth, [](BroadcastOnRtspAuthArgs) {
-            DebugL << "RTSP播放鉴权:" << args.getUrl() << " " << args.param_strs;
+            DebugL << "RTSP播放鉴权:" << args.getUrl() << " " << args.params;
             DebugL << "RTSP用户：" << user_name << (must_no_encrypt ? " Base64" : " MD5") << " 方式登录";
             string user = user_name;
             //假设我们异步读取数据库
@@ -134,14 +134,14 @@ void initEventListener() {
 
         //监听rtsp/rtmp推流事件，返回结果告知是否有推流权限
         NoticeCenter::Instance().addListener(nullptr, Broadcast::kBroadcastMediaPublish, [](BroadcastMediaPublishArgs) {
-            DebugL << "推流鉴权：" << args.getUrl() << " " << args.param_strs;
+            DebugL << "推流鉴权：" << args.getUrl() << " " << args.params;
             invoker("", ProtocolOption());//鉴权成功
             //invoker("this is auth failed message");//鉴权失败
         });
 
         //监听rtsp/rtsps/rtmp/http-flv播放事件，返回结果告知是否有播放权限(rtsp通过kBroadcastOnRtspAuth或此事件都可以实现鉴权)
         NoticeCenter::Instance().addListener(nullptr, Broadcast::kBroadcastMediaPlayed, [](BroadcastMediaPlayedArgs) {
-            DebugL << "播放鉴权:" << args.getUrl() << " " << args.param_strs;
+            DebugL << "播放鉴权:" << args.getUrl() << " " << args.params;
             invoker("");//鉴权成功
             //invoker("this is auth failed message");//鉴权失败
         });
@@ -183,13 +183,13 @@ void initEventListener() {
              * 你可以在这个事件触发时再去拉流，这样就可以实现按需拉流
              * 拉流成功后，ZLMediaKit会把其立即转发给播放器(最大等待时间约为5秒，如果5秒都未拉流成功，播放器会播放失败)
              */
-            DebugL << "未找到流事件:" << args.getUrl() << " " << args.param_strs;
+            DebugL << "未找到流事件:" << args.getUrl() << " " << args.params;
         });
 
 
         //监听播放或推流结束时消耗流量事件
         NoticeCenter::Instance().addListener(nullptr, Broadcast::kBroadcastFlowReport, [](BroadcastFlowReportArgs) {
-            DebugL << "播放器(推流器)断开连接事件:" << args.getUrl() << " " << args.param_strs << "\r\n使用流量:" << totalBytes << " bytes,连接时长:" << totalDuration << "秒";
+            DebugL << "播放器(推流器)断开连接事件:" << args.getUrl() << " " << args.params << "\r\n使用流量:" << totalBytes << " bytes,连接时长:" << totalDuration << "秒";
 
         });
 

webrtc/Sdp.cpp

@@ -23,10 +23,10 @@ namespace Rtc {
 const string kPreferredCodecA = RTC_FIELD "preferredCodecA";
 const string kPreferredCodecV = RTC_FIELD "preferredCodecV";
 static onceToken token([]() {
-    mINI::Instance()[kPreferredCodecA] = "PCMU,PCMA,opus,mpeg4-generic";
+    mINI::Instance()[kPreferredCodecA] = "PCMA,PCMU,opus,mpeg4-generic";
     mINI::Instance()[kPreferredCodecV] = "H264,H265,AV1,VP9,VP8";
 });
-}
+} // namespace Rtc
 
 using onCreateSdpItem = function<SdpItem::Ptr(const string &key, const string &value)>;
 static map<string, onCreateSdpItem, StrCaseCompare> sdpItemCreator;
@@ -73,9 +73,7 @@ public:
 
 class DirectionInterfaceImp : public SdpItem, public DirectionInterface {
 public:
-    DirectionInterfaceImp(RtpDirection direct){
+    DirectionInterfaceImp(RtpDirection direct) { direction = direct; }
-        direction = direct;
-    }
     const char *getKey() const override { return getRtpDirectionString(getDirection()); }
     RtpDirection getDirection() const override { return direction; }
 
@@ -717,8 +715,7 @@ void SdpAttrCandidate::parse(const string &str)  {
 
 string SdpAttrCandidate::toString() const {
     if (value.empty()) {
-        value = foundation + " " + to_string(component) + " " + transport + " " + to_string(priority) +
+        value = foundation + " " + to_string(component) + " " + transport + " " + to_string(priority) + " " + address + " " + to_string(port) + " typ " + type;
-                " " + address + " " + to_string(port) + " typ " + type;
         for (auto &pr : arr) {
             value += ' ';
             value += pr.first;
@@ -952,8 +949,7 @@ void RtcSession::loadFrom(const string &str) {
             if (fmtp_it != fmtp_map.end()) {
                 plan.fmtp = fmtp_it->second.fmtp;
             }
-            for (auto rtpfb_it = rtcpfb_map.find(pt);
+            for (auto rtpfb_it = rtcpfb_map.find(pt); rtpfb_it != rtcpfb_map.end() && rtpfb_it->second.pt == pt; ++rtpfb_it) {
-                 rtpfb_it != rtcpfb_map.end() && rtpfb_it->second.pt == pt; ++rtpfb_it) {
                 plan.rtcp_fb.emplace(rtpfb_it->second.rtcp_type);
             }
         }
@@ -986,8 +982,7 @@ void RtcSdpBase::toRtsp() {
             case 'a': {
                 auto attr = dynamic_pointer_cast<SdpAttr>(*it);
                 CHECK(attr);
-                if (!strcasecmp(attr->detail->getKey(), "rtpmap")
+                if (!strcasecmp(attr->detail->getKey(), "rtpmap") || !strcasecmp(attr->detail->getKey(), "fmtp")) {
-                    || !strcasecmp(attr->detail->getKey(), "fmtp")) {
                     ++it;
                     break;
                 }
@@ -1221,9 +1216,9 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{
             }
         }
     }
-    if(ice_lite)
+    if (ice_lite) {
         sdp.addAttr(std::make_shared<SdpCommon>("ice-lite"));
-    
+    }
     return ret;
 }
 
@@ -1427,16 +1422,14 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
             preferred_codec = s_preferred_codec;
 
             rtcp_fb = { SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb };
-            extmap = {
+            extmap = { { RtpExtType::ssrc_audio_level, RtpDirection::sendrecv },
-                    {RtpExtType::ssrc_audio_level,            RtpDirection::sendrecv},
                        { RtpExtType::csrc_audio_level, RtpDirection::sendrecv },
                        { RtpExtType::abs_send_time, RtpDirection::sendrecv },
                        { RtpExtType::transport_cc, RtpDirection::sendrecv },
                        // rtx重传rtp时，忽略sdes_mid类型的rtp ext,实测发现Firefox在接收rtx时，如果存在sdes_mid的ext,将导致无法播放
                        //{RtpExtType::sdes_mid,RtpDirection::sendrecv},
                        { RtpExtType::sdes_rtp_stream_id, RtpDirection::sendrecv },
-                    {RtpExtType::sdes_repaired_rtp_stream_id, RtpDirection::sendrecv}
+                       { RtpExtType::sdes_repaired_rtp_stream_id, RtpDirection::sendrecv } };
-            };
             break;
         }
         case TrackVideo: {
@@ -1446,8 +1439,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
             preferred_codec = s_preferred_codec;
 
             rtcp_fb = { SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb, "nack", "ccm fir", "nack pli" };
-            extmap = {
+            extmap = { { RtpExtType::abs_send_time, RtpDirection::sendrecv },
-                    {RtpExtType::abs_send_time,               RtpDirection::sendrecv},
                        { RtpExtType::transport_cc, RtpDirection::sendrecv },
                        // rtx重传rtp时，忽略sdes_mid类型的rtp ext,实测发现Firefox在接收rtx时，如果存在sdes_mid的ext,将导致无法播放
                        //{RtpExtType::sdes_mid,RtpDirection::sendrecv},
@@ -1460,8 +1452,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
                        // 手机端推webrtc 会带有旋转角度，rtc协议能正常播放 其他协议拉流画面旋转
                        //{RtpExtType::video_orientation,           RtpDirection::sendrecv},
                        { RtpExtType::toffset, RtpDirection::sendrecv },
-                    {RtpExtType::framemarking,                RtpDirection::sendrecv}
+                       { RtpExtType::framemarking, RtpDirection::sendrecv } };
-            };
             break;
         }
         case TrackApplication: {
@@ -1471,8 +1462,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
     }
 }
 
-void RtcConfigure::setDefaultSetting(string ice_ufrag, string ice_pwd, RtpDirection direction,
+void RtcConfigure::setDefaultSetting(string ice_ufrag, string ice_pwd, RtpDirection direction, const SdpAttrFingerprint &fingerprint) {
-                                     const SdpAttrFingerprint &fingerprint) {
     video.setDefaultSetting(TrackVideo);
     audio.setDefaultSetting(TrackAudio);
     application.setDefaultSetting(TrackApplication);
@@ -1683,8 +1673,7 @@ RETRY:
         answer_media.role = mathDtlsRole(offer_media.role);
 
         // 如果codec匹配失败，那么禁用该track
-        answer_media.direction = check_codec ? matchDirection(offer_media.direction, configure.direction)
+        answer_media.direction = check_codec ? matchDirection(offer_media.direction, configure.direction) : RtpDirection::inactive;
-                                             : RtpDirection::inactive;
         if (answer_media.direction == RtpDirection::invalid) {
             continue;
         }

webrtc/Sdp.h

@@ -77,11 +77,7 @@ enum class DtlsRole {
     actpass,
 };
 
-enum class SdpType {
+enum class SdpType { invalid = -1, offer, answer };
-    invalid = -1,
-    offer,
-    answer
-};
 
 DtlsRole getDtlsRole(const std::string &str);
 const char *getDtlsRoleString(DtlsRole role);
@@ -92,17 +88,11 @@ class SdpItem {
 public:
     using Ptr = std::shared_ptr<SdpItem>;
     virtual ~SdpItem() = default;
-    virtual void parse(const std::string &str) {
+    virtual void parse(const std::string &str) { value = str; }
-        value  = str;
+    virtual std::string toString() const { return value; }
-    }
-    virtual std::string toString() const {
-        return value;
-    }
     virtual const char *getKey() const = 0;
 
-    void reset() {
+    void reset() { value.clear(); }
-        value.clear();
-    }
 
 protected:
     mutable std::string value;
@@ -259,9 +249,7 @@ public:
     void parse(const std::string &str) override;
     std::string toString() const override;
     const char *getKey() const override { return "msid-semantic"; }
-    bool empty() const {
+    bool empty() const { return msid.empty(); }
-        return msid.empty();
-    }
 };
 
 class SdpAttrRtcp : public SdpItem {
@@ -271,12 +259,11 @@ public:
     std::string nettype { "IN" };
     std::string addrtype { "IP4" };
     std::string address { "0.0.0.0" };
-    void parse(const std::string &str) override;;
+    void parse(const std::string &str) override;
+    ;
     std::string toString() const override;
     const char *getKey() const override { return "rtcp"; }
-    bool empty() const {
+    bool empty() const { return address.empty() || !port; }
-        return address.empty() || !port;
-    }
 };
 
 class SdpAttrIceUfrag : public SdpItem {
@@ -364,9 +351,9 @@ public:
     // a=rtcp-fb:98 nack pli
     // a=rtcp-fb:120 nack 支持 nack 重传，nack (Negative-Acknowledgment) 。
     // a=rtcp-fb:120 nack pli 支持 nack 关键帧重传，PLI (Picture Loss Indication) 。
-    //a=rtcp-fb:120 ccm fir 支持编码层关键帧请求，CCM (Codec Control Message)，FIR (Full Intra Request )，通常与 nack pli 有同样的效果，但是 nack pli 是用于重传时的关键帧请求。
+    // a=rtcp-fb:120 ccm fir 支持编码层关键帧请求，CCM (Codec Control Message)，FIR (Full Intra Request )，通常与 nack pli 有同样的效果，但是 nack pli
-    //a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。
+    // 是用于重传时的关键帧请求。 a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。 a=rtcp-fb:120 transport-cc 支持 TCC (Transport
-    //a=rtcp-fb:120 transport-cc 支持 TCC (Transport Congest Control) 。
+    // Congest Control) 。
     uint8_t pt;
     std::string rtcp_type;
     void parse(const std::string &str) override;

webrtc/WebRtcEchoTest.cpp

@@ -45,7 +45,7 @@ void WebRtcEchoTest::onCheckSdp(SdpType type, RtcSession &sdp) {
         for (auto &m : sdp.media) {
             for (auto &ssrc : m.rtp_rtx_ssrc) {
                 if (!ssrc.msid.empty()) {
-                    ssrc.msid = "zlmediakit msid";
+                    ssrc.msid = "zlmediakit-mslabel zlmediakit-label-" + m.mid;
                 }
             }
         }

webrtc/WebRtcEchoTest.h

@@ -27,7 +27,6 @@ protected:
     void onRtp(const char *buf, size_t len, uint64_t stamp_ms) override;
     void onRtcp(const char *buf, size_t len) override;
 
-    void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
     void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) override {};
     void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) override {};
 

webrtc/WebRtcPlayer.cpp

@@ -17,9 +17,8 @@ namespace mediakit {
 
 WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
                                        const RtspMediaSource::Ptr &src,
-                                       const MediaInfo &info,
+                                       const MediaInfo &info) {
-                                       bool preferred_tcp) {
+    WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info), [](WebRtcPlayer *ptr) {
-    WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info, preferred_tcp), [](WebRtcPlayer *ptr) {
         ptr->onDestory();
         delete ptr;
     });
@@ -29,8 +28,7 @@ WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
 
 WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
                            const RtspMediaSource::Ptr &src,
-                           const MediaInfo &info,
+                           const MediaInfo &info) : WebRtcTransportImp(poller) {
-                           bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
     _media_info = info;
     _play_src = src;
     CHECK(src);

webrtc/WebRtcPlayer.h

@@ -19,7 +19,7 @@ namespace mediakit {
 class WebRtcPlayer : public WebRtcTransportImp {
 public:
     using Ptr = std::shared_ptr<WebRtcPlayer>;
-    static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp = false);
+    static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
     MediaInfo getMediaInfo() { return _media_info; }
 
 protected:
@@ -27,10 +27,9 @@ protected:
     void onStartWebRTC() override;
     void onDestory() override;
     void onRtcConfigure(RtcConfigure &configure) const override;
-    void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
 
 private:
-    WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp);
+    WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
 
 private:
     //媒体相关元数据

webrtc/WebRtcPusher.cpp

@@ -20,9 +20,8 @@ WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
                                        const RtspMediaSource::Ptr &src,
                                        const std::shared_ptr<void> &ownership,
                                        const MediaInfo &info,
-                                       const ProtocolOption &option,
+                                       const ProtocolOption &option) {
-                                       bool preferred_tcp) {
+    WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
-    WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option,preferred_tcp), [](WebRtcPusher *ptr) {
         ptr->onDestory();
         delete ptr;
     });
@@ -34,8 +33,7 @@ WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
                            const RtspMediaSource::Ptr &src,
                            const std::shared_ptr<void> &ownership,
                            const MediaInfo &info,
-                           const ProtocolOption &option,
+                           const ProtocolOption &option) : WebRtcTransportImp(poller) {
-                           bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
     _media_info = info;
     _push_src = src;
     _push_src_ownership = ownership;

webrtc/WebRtcPusher.h

@@ -20,8 +20,7 @@ class WebRtcPusher : public WebRtcTransportImp, public MediaSourceEvent {
 public:
     using Ptr = std::shared_ptr<WebRtcPusher>;
     static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src,
-                      const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp = false);
+                      const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
-
 
 protected:
     ///////WebRtcTransportImp override///////
@@ -53,7 +52,7 @@ protected:
 
 private:
     WebRtcPusher(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src,
-                 const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp);
+                 const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
 
 private:
     bool _simulcast = false;

webrtc/WebRtcTransport.cpp

@@ -378,6 +378,12 @@ void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote) {
 }
 
 void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
+    SdpAttrFingerprint fingerprint;
+    fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
+    fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
+    configure.setDefaultSetting(
+            _ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
+
     // 开启remb后关闭twcc，因为开启twcc后remb无效
     GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
     configure.enableTWCC(!remb_bit_rate);
@@ -407,12 +413,7 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer) {
         setRemoteDtlsFingerprint(*_offer_sdp);
 
         //// sdp 配置 ////
-        SdpAttrFingerprint fingerprint;
-        fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
-        fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
         RtcConfigure configure;
-        configure.setDefaultSetting(
-            _ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
         onRtcConfigure(configure);
 
         //// 生成answer sdp ////
@@ -431,10 +432,6 @@ static bool isDtls(char *buf) {
     return ((*buf > 19) && (*buf < 64));
 }
 
-static string getPeerAddress(RTC::TransportTuple *tuple) {
-    return tuple->get_peer_ip();
-}
-
 void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
     if (RTC::StunPacket::IsStun((const uint8_t *)buf, len)) {
         std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *)buf, len));
@@ -451,7 +448,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
     }
     if (isRtp(buf, len)) {
         if (!_srtp_session_recv) {
-            WarnL << "received rtp packet when dtls not completed from:" << getPeerAddress(tuple);
+            WarnL << "received rtp packet when dtls not completed from:" << tuple->get_peer_ip();
             return;
         }
         if (_srtp_session_recv->DecryptSrtp((uint8_t *)buf, &len)) {
@@ -461,7 +458,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
     }
     if (isRtcp(buf, len)) {
         if (!_srtp_session_recv) {
-            WarnL << "received rtcp packet when dtls not completed from:" << getPeerAddress(tuple);
+            WarnL << "received rtcp packet when dtls not completed from:" << tuple->get_peer_ip();
             return;
         }
         if (_srtp_session_recv->DecryptSrtcp((uint8_t *)buf, &len)) {
@@ -533,8 +530,7 @@ void WebRtcTransportImp::OnDtlsTransportApplicationDataReceived(const RTC::DtlsT
 #endif
 }
 
-WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp)
+WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
-    : WebRtcTransport(poller), _preferred_tcp(preferred_tcp) {
     InfoL << getIdentifier();
 }
 
@@ -674,7 +670,7 @@ void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
     });
     for (auto &m : sdp.media) {
         m.addr.reset();
-        m.addr.address = extern_ips.empty() ? _localIp.empty() ? SockUtil::get_local_ip() : _localIp : extern_ips[0];
+        m.addr.address = extern_ips.empty() ? _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip : extern_ips[0];
         m.rtcp_addr.reset();
         m.rtcp_addr.address = m.addr.address;
 
@@ -769,7 +765,7 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
         return ret;
     });
     if (extern_ips.empty()) {
-        std::string local_ip = _localIp.empty() ? SockUtil::get_local_ip() : _localIp;
+        std::string local_ip = _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip;
         if (local_udp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_udp_port, 120, "udp")); }
         if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, _preferred_tcp ? 125 : 115, "tcp")); }
     } else {
@@ -783,12 +779,16 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
     }
 }
 
-void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
+void WebRtcTransportImp::setPreferredTcp(bool flag) {
-    _cands = std::move(cands);
+    _preferred_tcp = flag;
 }
 
-void WebRtcTransportImp::setLocalIp(const std::string &localIp) {
+void WebRtcTransportImp::setLocalIp(std::string local_ip) {
-    _localIp = localIp;
+    _local_ip = std::move(local_ip);
+}
+
+void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
+    _cands = std::move(cands);
 }
 
 ///////////////////////////////////////////////////////////////////
@@ -1278,21 +1278,14 @@ void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
     _map_creator[type] = std::move(cb);
 }
 
-std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer) {
-    return const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer);
-}
-
-void setLocalIp(const WebRtcInterface& exchanger, const std::string& localIp) {
-    return const_cast<WebRtcInterface &>(exchanger).setLocalIp(localIp);
-}
 
 void WebRtcPluginManager::setListener(Listener cb) {
     lock_guard<mutex> lck(_mtx_creator);
     _listener = std::move(cb);
 }
 
-void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateRtc &cb_in) {
+void WebRtcPluginManager::negotiateSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateWebRtc &cb_in) {
-    onCreateRtc cb;
+    onCreateWebRtc cb;
     lock_guard<mutex> lck(_mtx_creator);
     if (_listener) {
         auto listener = _listener;
@@ -1308,21 +1301,19 @@ void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, cons
 
     auto it = _map_creator.find(type);
     if (it == _map_creator.end()) {
-        cb(WebRtcException(SockException(Err_other, "the type can not supported")));
+        cb_in(WebRtcException(SockException(Err_other, "the type can not supported")));
         return;
     }
     it->second(sender, args, cb);
 }
 
-void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
+void echo_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
     cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
 }
 
-void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
+void push_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
     MediaInfo info(args["url"]);
-    bool preferred_tcp = args["preferred_tcp"];
+    Broadcast::PublishAuthInvoker invoker = [cb, info](const string &err, const ProtocolOption &option) mutable {
-
-    Broadcast::PublishAuthInvoker invoker = [cb, info, preferred_tcp](const string &err, const ProtocolOption &option) mutable {
         if (!err.empty()) {
             cb(WebRtcException(SockException(Err_other, err)));
             return;
@@ -1361,7 +1352,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
             push_src_ownership = push_src->getOwnership();
             push_src->setProtocolOption(option);
         }
-        auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option, preferred_tcp);
+        auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
         push_src->setListener(rtc);
         cb(*rtc);
     };
@@ -1374,12 +1365,10 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
     }
 }
 
-void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
+void play_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
     MediaInfo info(args["url"]);
-    bool preferred_tcp = args["preferred_tcp"];
-
     auto session_ptr = static_pointer_cast<Session>(sender.shared_from_this());
-    Broadcast::AuthInvoker invoker = [cb, info, session_ptr, preferred_tcp](const string &err) mutable {
+    Broadcast::AuthInvoker invoker = [cb, info, session_ptr](const string &err) mutable {
         if (!err.empty()) {
             cb(WebRtcException(SockException(Err_other, err)));
             return;
@@ -1395,7 +1384,7 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
             }
             // 还原成rtc，目的是为了hook时识别哪种播放协议
             info.schema = "rtc";
-            auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info, preferred_tcp);
+            auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
             cb(*rtc);
         });
     };
@@ -1408,14 +1397,35 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
     }
 }
 
-static void set_webrtc_cands(const WebRtcArgs &args, const WebRtcInterface &rtc) {
+static void setWebRtcArgs(const WebRtcArgs &args, WebRtcInterface &rtc) {
+    {
+        static auto is_vaild_ip = [](const std::string &ip) -> bool {
+            int a, b, c, d;
+            return sscanf(ip.c_str(), "%d.%d.%d.%d", &a, &b, &c, &d) == 4;
+        };
+        std::string host = args["Host"];
+        if (!host.empty()) {
+            auto local_ip = host.substr(0, host.find(':'));
+            if (!is_vaild_ip(local_ip) || local_ip == "127.0.0.1") {
+                local_ip = "";
+            }
+            rtc.setLocalIp(std::move(local_ip));
+        }
+    }
+
+    bool preferred_tcp = args["preferred_tcp"];
+    {
+        rtc.setPreferredTcp(preferred_tcp);
+    }
+
+    {
         vector<SdpAttrCandidate> cands;
         {
             auto cand_str = trim(args["cand_udp"]);
             auto ip_port = toolkit::split(cand_str, ":");
             if (ip_port.size() == 2) {
                 // udp优先
-            auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), 120, "udp");
+                auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 100 : 120, "udp");
                 cands.emplace_back(std::move(*ice_cand));
             }
         }
@@ -1424,23 +1434,26 @@ static void set_webrtc_cands(const WebRtcArgs &args, const WebRtcInterface &rtc)
             auto ip_port = toolkit::split(cand_str, ":");
             if (ip_port.size() == 2) {
                 // tcp模式
-            auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), 100, "tcp");
+                auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 120 : 100, "tcp");
                 cands.emplace_back(std::move(*ice_cand));
             }
         }
         if (!cands.empty()) {
             // udp优先
-        const_cast<WebRtcInterface &>(rtc).setIceCandidate(std::move(cands));
+            rtc.setIceCandidate(std::move(cands));
+        }
     }
 }
 
 static onceToken s_rtc_auto_register([]() {
+#if !defined (NDEBUG)
+    // debug模式才开启echo插件
     WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
+#endif
     WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
     WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
-
     WebRtcPluginManager::Instance().setListener([](Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc) {
-        set_webrtc_cands(args, rtc);
+        setWebRtcArgs(args, const_cast<WebRtcInterface&>(rtc));
     });
 });
 

webrtc/WebRtcTransport.h

@@ -42,13 +42,10 @@ public:
     virtual const std::string& getIdentifier() const = 0;
     virtual const std::string& deleteRandStr() const { static std::string s_null; return s_null; }
     virtual void setIceCandidate(std::vector<SdpAttrCandidate> cands) {}
-    virtual void setLocalIp(const std::string &localIp) {}
+    virtual void setLocalIp(std::string localIp) {}
+    virtual void setPreferredTcp(bool flag) {}
 };
 
-std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer);
-
-void setLocalIp(const WebRtcInterface &exchanger, const std::string &localIp);
-
 class WebRtcException : public WebRtcInterface {
 public:
     WebRtcException(const SockException &ex) : _ex(ex) {};
@@ -88,7 +85,7 @@ public:
      * @param offer offer sdp
      * @return answer sdp
      */
-    std::string getAnswerSdp(const std::string &offer) override;
+    std::string getAnswerSdp(const std::string &offer) override final;
 
     /**
      * 获取对象唯一id
@@ -252,14 +249,16 @@ public:
     void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
 
     void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
-    void setIceCandidate(std::vector<SdpAttrCandidate> cands) override;
     void removeTuple(RTC::TransportTuple* tuple);
     void safeShutdown(const SockException &ex);
 
-    void setLocalIp(const std::string &localIp) override;
+    void setPreferredTcp(bool flag) override;
+    void setLocalIp(std::string local_ip) override;
+    void setIceCandidate(std::vector<SdpAttrCandidate> cands) override;
+
 protected:
     void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override;
-    WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp = false);
+    WebRtcTransportImp(const EventPoller::Ptr &poller);
     void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
     void onStartWebRTC() override;
     void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
@@ -273,7 +272,7 @@ protected:
     void onCreate() override;
     void onDestory() override;
     void onShutdown(const SockException &ex) override;
-    virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) = 0;
+    virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) {}
     void updateTicker();
     float getLossRate(TrackType type);
     void onRtcpBye() override;
@@ -289,7 +288,7 @@ private:
     void onCheckAnswer(RtcSession &sdp);
 
 private:
-    bool _preferred_tcp;
+    bool _preferred_tcp = false;
     uint16_t _rtx_seq[2] = {0, 0};
     //用掉的总流量
     uint64_t _bytes_usage = 0;
@@ -310,8 +309,8 @@ private:
     //根据接收rtp的pt获取相关信息
     std::unordered_map<uint8_t/*pt*/, std::unique_ptr<WrappedMediaTrack>> _pt_to_track;
     std::vector<SdpAttrCandidate> _cands;
-    //源访问的hostip
+    //http访问时的host ip
-    std::string _localIp;
+    std::string _local_ip;
 };
 
 class WebRtcTransportManager {
@@ -333,21 +332,20 @@ private:
 class WebRtcArgs : public std::enable_shared_from_this<WebRtcArgs> {
 public:
     virtual ~WebRtcArgs() = default;
-
     virtual variant operator[](const std::string &key) const = 0;
 };
 
+using onCreateWebRtc = std::function<void(const WebRtcInterface &rtc)>;
 class WebRtcPluginManager {
 public:
-    using onCreateRtc = std::function<void(const WebRtcInterface &rtc)>;
+    using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb)>;
-    using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateRtc &cb)>;
     using Listener = std::function<void(Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc)>;
 
     static WebRtcPluginManager &Instance();
 
     void registerPlugin(const std::string &type, Plugin cb);
-    void getAnswerSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateRtc &cb);
     void setListener(Listener cb);
+    void negotiateSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateWebRtc &cb);
 
 private:
     WebRtcPluginManager() = default;