Merge branch 'master' into dev

.github/ISSUE_TEMPLATE/bug.md

@@ -20,14 +20,14 @@ assignees: ''
  * https://docs.github.com/en/get-started/writing-on-github/getting-started-with-writing-and-formatting-on-github/basic-writing-and-formatting-syntax
  -->
 
-### 现象描述
+## 现象描述
 
 <!--
  在使用什么功能产生的问题? 其异常表现是什么?
  如: 在测试 WebRTC 功能时, 使用 Chrome 浏览器访问 ZLMediait 自带网页播放 FFmpeg 以 RTSP 协议推送的图像有卡顿/花屏.
  -->
 
-### 如何复现?
+## 如何复现?
 
 <!--
   明确的复现步骤对快速解决问题极有帮助.
@@ -37,7 +37,7 @@ assignees: ''
     1. 期望 ..., 结果 ...
   -->
 
-### 相关日志或截图
+## 相关日志或截图
 
 <!--
   由于日志通长较长, 建议将日志信息填写到下面的 "日志内容..."
@@ -50,11 +50,14 @@ assignees: ''
 <details>
 <summary>展开查看详细日志</summary>
 <pre>
-日志内容...
+
+```
+#详细日志粘在这里!
+```
 </pre>
 </details>
 
-### 配置
+## 配置
 
 <!--
   部分常见问题是由于配置错误导致的, 建议仔细阅读配置文件中的注释信息
@@ -65,11 +68,14 @@ assignees: ''
 <details>
 <summary>展开查看详细配置</summary>
 <pre>
-配置内容...
+
+```ini
+#config.ini内容粘在这里!
+```
 </pre>
 </details>
 
-### 各种环境信息
+## 各种环境信息
 
 <!--
   请填写相关环境信息, 详细的环境信息有助于快速复现定位问题.
@@ -82,4 +88,8 @@ assignees: ''
 * **代码提交记录/git commit hash**:
 * **操作系统及版本**:
 * **硬件信息**:
+* **crash backtrace**:
+```
+#崩溃信息backtrace粘贴至此
+```
 * **其他需要补充的信息**:

.github/ISSUE_TEMPLATE/compile.md

@@ -20,7 +20,7 @@ assignees: ''
  * https://docs.github.com/en/get-started/writing-on-github/getting-started-with-writing-and-formatting-on-github/basic-writing-and-formatting-syntax
  -->
 
-### 相关日志及环境信息
+## 相关日志及环境信息
 
 <!--
   由于编译日志通长较长, 建议将日志信息填写到下面 `````` block 内，或者上传日志文件
@@ -41,7 +41,7 @@ assignees: ''
 
 编译目录下的 `CMakeCache.txt` 文件内容，请直接上传为附件。
 
-### 各种环境信息
+## 各种环境信息
 
 <!--
   请填写相关环境信息, 详细的环境信息有助于快速复现定位问题.

.github/ISSUE_TEMPLATE/config.yml

@@ -0,0 +1,6 @@
+blank_issues_enabled: false
+
+contact_links:
+  - name: 技术咨询
+    url: https://t.zsxq.com/FcVK5
+    about: 请在知识星球发起技术咨询

.github/ISSUE_TEMPLATE/feature.md

@@ -7,8 +7,8 @@ assignees: ''
 
 ---
 
-**描述该功能的用处，可以提供相关资料描述该功能**
+## 描述该功能的用处，可以提供相关资料描述该功能
 
-**该功能是否用于改进项目缺陷，如果是，请描述现有缺陷**
+## 该功能是否用于改进项目缺陷，如果是，请描述现有缺陷
 
-**描述你期望实现该功能的方式和最终效果**
+## 描述你期望实现该功能的方式和最终效果

.github/ISSUE_TEMPLATE/question.md

@@ -1,19 +0,0 @@
----
-name: 技术咨询
-about: 使用咨询、技术咨询等
-title: "[技术咨询] 咨询描述(必填)"
-labels: 技术咨询
-assignees: ''
-
----
-
-**咨询的功能模块**
-- 请描述您想咨询zlmediakit的哪部分功能
-
-**咨询的具体内容和问题**
-- 此处展开您咨询内容的描述
-
-**注意事项**
-- 技术咨询前请先认真阅读readme, [wiki](https://github.com/ZLMediaKit/ZLMediaKit/wiki),如有必要，您也可以同时搜索已经答复的issue，如果没找到答案才在此提issue
-
-- 技术咨询不属于bug缺陷，要求用户先star(收藏)本项目，否则会直接关闭issue

.github/ISSUE_TEMPLATE/requirement.md

@@ -1,8 +0,0 @@
----
-name: issue创建要求
-about: 不符合模板要求不便定位问题，可能会被管理员直接关闭
-title: ""
-labels: ''
-assignees: ''
-
----

.github/workflows/android.yml

@@ -23,3 +23,35 @@ jobs:
 
     - name: 编译
       run: cd Android && ./gradlew build
+
+    - name: 设置环境变量
+      run: |
+        echo "BRANCH=$(echo ${GITHUB_REF#refs/heads/} | tr -s "/\?%*:|\"<>" "_")" >> $GITHUB_ENV  
+        echo "BRANCH2=$(echo ${GITHUB_REF#refs/heads/} )" >> $GITHUB_ENV    
+        echo "DATE=$(date +%Y-%m-%d)" >> $GITHUB_ENV
+
+    - name: 打包二进制
+      id: upload
+      uses: actions/upload-artifact@v4
+      with:
+        name: ${{ github.workflow }}_${{ env.BRANCH }}_${{ env.DATE }}
+        path: Android/app/build/outputs/apk/debug/*
+        if-no-files-found: error
+        retention-days: 90
+
+    - name: issue评论
+      if: github.event_name != 'pull_request' && github.ref != 'refs/heads/feature/test'
+      uses: actions/github-script@v7
+      with:
+        github-token: ${{ secrets.GITHUB_TOKEN }}
+        script: |
+          github.rest.issues.createComment({
+            issue_number: 483,
+            owner: context.repo.owner,
+            repo: context.repo.repo,
+            body: '- 下载地址: [${{ github.workflow }}_${{ env.BRANCH }}_${{ env.DATE }}](${{ steps.upload.outputs.artifact-url }})\n'
+              + '- 分支: ${{ env.BRANCH2 }}\n' 
+              + '- git hash: ${{ github.sha }} \n' 
+              + '- 编译日期: ${{ env.DATE }}\n' 
+              + '- 打包ci名: ${{ github.workflow }}\n' 
+          })

.github/workflows/docker.yml

@@ -7,12 +7,6 @@ on:
       - "feature/*"
       - "release/*"
 
-  pull_request:
-    branches:
-      - "master"
-      - "feature/*"
-      - "release/*"
-
 env:
   # Use docker.io for Docker Hub if empty
   REGISTRY: docker.io
@@ -21,7 +15,7 @@ env:
 jobs:
   build:
 
-    runs-on: ubuntu-latest
+    runs-on: ubuntu-20.04
     permissions:
       contents: read
       packages: write
@@ -39,7 +33,6 @@ jobs:
       # Install the cosign tool except on PR
       # https://github.com/sigstore/cosign-installer
       - name: Install cosign
-        if: github.event_name != 'pull_request'
         uses: sigstore/cosign-installer@d572c9c13673d2e0a26fabf90b5748f36886883f
 
       - name: Set up QEMU
@@ -53,7 +46,6 @@ jobs:
       # Login against a Docker registry except on PR
       # https://github.com/docker/login-action
       - name: Log into registry ${{ env.REGISTRY }}
-        if: github.event_name != 'pull_request'
         uses: docker/login-action@28218f9b04b4f3f62068d7b6ce6ca5b26e35336c
         with:
           registry: ${{ env.REGISTRY }}
@@ -71,6 +63,7 @@ jobs:
       # Build and push Docker image with Buildx (don't push on PR)
       # https://github.com/docker/build-push-action
       - name: Build and push Docker image
+        if: github.event_name != 'pull_request' && github.ref != 'refs/heads/feature/test'
         id: build-and-push
         uses: docker/build-push-action@ac9327eae2b366085ac7f6a2d02df8aa8ead720a
         with:

.github/workflows/issue_lint.yml

@@ -0,0 +1,51 @@
+name: issue_lint
+
+on:
+  issues:
+    types: [opened]
+
+jobs:
+  issue_lint:
+    runs-on: ubuntu-22.04
+
+    steps:
+      - uses: actions/checkout@v3
+
+      - uses: actions/github-script@v6
+        with:
+          github-token: ${{ secrets.GITHUB_TOKEN }}
+          script: |
+            const fs = require('fs').promises;
+            
+            const getTitles = (str) => (
+              [...str.matchAll(/^## (.*)/gm)].map((m) => m[0])
+            );
+            
+            const titles = getTitles(context.payload.issue.body);
+            
+            for (let file of await fs.readdir('.github/ISSUE_TEMPLATE')) {
+              if (!file.endsWith('.md')) {
+                continue;
+              }
+            
+              const template = await fs.readFile(`.github/ISSUE_TEMPLATE/${file}`, 'utf-8');
+              const templateTitles = getTitles(template);
+            
+              if (templateTitles.every((title) => titles.includes(title))) {
+                process.exit(0);
+              }
+            }
+            
+            await github.rest.issues.createComment({
+              owner: context.issue.owner,
+              repo: context.issue.repo,
+              issue_number: context.issue.number,
+              body: '此issue由于不符合模板规范已经自动关闭，请重新按照模板规范确保包含模板中所有章节标题再提交\n',
+            });
+            
+            await github.rest.issues.update({
+              owner: context.issue.owner,
+              repo: context.issue.repo,
+              issue_number: context.issue.number,
+              state: 'closed',
+            });

.github/workflows/linux.yml

@@ -13,9 +13,6 @@ jobs:
     - name: 下载submodule源码
       run: mv -f .gitmodules_github .gitmodules && git submodule sync && git submodule update --init
 
-    - name: apt-get安装依赖库(非必选)
-      run: sudo apt-get update && sudo apt-get install -y cmake libssl-dev libsdl-dev libavcodec-dev libavutil-dev libswscale-dev libresample-dev libusrsctp-dev
-
     - name: 下载 SRTP
       uses: actions/checkout@v2
       with:
@@ -24,13 +21,64 @@ jobs:
         ref: v2.3.0
         path: 3rdpart/libsrtp
 
-    - name: 编译 SRTP
+    - name: 启动 Docker 容器, 在Docker 容器中执行脚本
-      run: cd 3rdpart/libsrtp && ./configure --enable-openssl && make -j4 && sudo make install
+      run: |
+        docker pull centos:7 
+        docker run -v $(pwd):/root -w /root --rm centos:7 sh -c "
+          set -x
+          yum install -y git wget gcc gcc-c++ make unzip ffmpeg-devel libavutil-devel libswscale-devel libresample-devel usrsctp-devel
           
-    - name: 编译
+          wget https://github.com/openssl/openssl/archive/refs/heads/OpenSSL_1_1_1-stable.zip
-      run: mkdir -p linux_build && cd linux_build && cmake .. -DENABLE_WEBRTC=true -DENABLE_FFMPEG=true && make -j $(nproc)
+          unzip OpenSSL_1_1_1-stable.zip
+          cd openssl-OpenSSL_1_1_1-stable
+          ./config no-shared --prefix=/root/release
+          make -j $(nproc)
+          make install
+          cd ..
         
-    - name: 运行MediaServer
+          wget https://github.com/Kitware/CMake/releases/download/v3.29.5/cmake-3.29.5.tar.gz
-      run: pwd && cd release/linux/Debug && sudo ./MediaServer -d &
+          tar -xvf cmake-3.29.5.tar.gz
+          cd cmake-3.29.5
+          OPENSSL_ROOT_DIR=/root/release ./configure
+          make -j $(nproc)
+          make install
+          cd ..
         
+          cd 3rdpart/libsrtp && ./configure --enable-openssl  --with-openssl-dir=/root/release && make -j $(nproc) && make install
+          cd ../../
         
+          mkdir -p linux_build && cd linux_build && cmake .. -DOPENSSL_ROOT_DIR=/root/release -DCMAKE_BUILD_TYPE=Release -DENABLE_FFMPEG=true && make -j $(nproc)
+        "
+
+    - name: 设置环境变量
+      run: |
+        echo "BRANCH=$(echo ${GITHUB_REF#refs/heads/} | tr -s "/\?%*:|\"<>" "_")" >> $GITHUB_ENV  
+        echo "BRANCH2=$(echo ${GITHUB_REF#refs/heads/} )" >> $GITHUB_ENV    
+        echo "DATE=$(date +%Y-%m-%d)" >> $GITHUB_ENV
+
+    - name: 打包二进制
+      id: upload
+      uses: actions/upload-artifact@v4
+      with:
+        name: ${{ github.workflow }}_${{ env.BRANCH }}_${{ env.DATE }}
+        path: release/*
+        if-no-files-found: error
+        retention-days: 90
+
+    - name: issue评论
+      if: github.event_name != 'pull_request' && github.ref != 'refs/heads/feature/test'
+      uses: actions/github-script@v7
+      with:
+        github-token: ${{ secrets.GITHUB_TOKEN }}
+        script: |
+          github.rest.issues.createComment({
+            issue_number: 483,
+            owner: context.repo.owner,
+            repo: context.repo.repo,
+            body: '- 下载地址: [${{ github.workflow }}_${{ env.BRANCH }}_${{ env.DATE }}](${{ steps.upload.outputs.artifact-url }})\n'
+              + '- 分支: ${{ env.BRANCH2 }}\n' 
+              + '- git hash: ${{ github.sha }} \n' 
+              + '- 编译日期: ${{ env.DATE }}\n' 
+              + '- 打包ci名: ${{ github.workflow }}\n' 
+              + '- 说明: 本二进制在centos7(x64)上编译，请确保您的机器系统不低于此版本，并提前`sudo yum check-update && sudo yum install -y openssl-devel ffmpeg-devel libavutil-devel libswscale-devel libresample-devel usrsctp-devel`安装依赖项\n' 
+          })

.github/workflows/macos.yml

@@ -13,27 +13,52 @@ jobs:
     - name: 下载submodule源码
       run: mv -f .gitmodules_github .gitmodules && git submodule sync && git submodule update --init
 
-#    - name: 安装brew
+    - name: 配置 vcpkg
-#      run: ruby -e "$(curl -fsSL https://raw.githubusercontent.com/Homebrew/install/master/install)"
+      uses: lukka/run-vcpkg@v7
-#
+      with:
-#    - name: brew安装依赖库(非必选)
+        vcpkgDirectory: '${{github.workspace}}/vcpkg'
-#      run: brew update && brew install cmake openssl sdl2 ffmpeg
+        vcpkgTriplet: arm64-osx
-
+        # 2024.06.01
-#    - name: 下载 SRTP
+        vcpkgGitCommitId: '47364fbc300756f64f7876b549d9422d5f3ec0d3'
-#      uses: actions/checkout@v2
+        vcpkgArguments: 'openssl libsrtp[openssl]'
-#      with:
-#        repository: cisco/libsrtp
-#        fetch-depth: 1
-#        ref: v2.3.0
-#        path: 3rdpart/libsrtp
-#
-#    - name: 编译 SRTP
-#      run: cd 3rdpart/libsrtp && ./configure --enable-openssl && make -j4 && sudo make install
 
     - name: 编译
-      run: mkdir -p build && cd build && cmake .. &&  make -j $(nproc)
+      uses: lukka/run-cmake@v3
+      with:
+        useVcpkgToolchainFile: true
+        buildDirectory: '${{github.workspace}}/build'
+        cmakeAppendedArgs: ''
+        cmakeBuildType: 'Release'
 
-    - name: 运行MediaServer
+    - name: 设置环境变量
-      run: pwd && cd release/darwin/Debug && sudo ./MediaServer -d &
+      run: |
+        echo "BRANCH=$(echo ${GITHUB_REF#refs/heads/} | tr -s "/\?%*:|\"<>" "_")" >> $GITHUB_ENV  
+        echo "BRANCH2=$(echo ${GITHUB_REF#refs/heads/} )" >> $GITHUB_ENV    
+        echo "DATE=$(date +%Y-%m-%d)" >> $GITHUB_ENV
 
+    - name: 打包二进制
+      id: upload
+      uses: actions/upload-artifact@v4
+      with:
+        name: ${{ github.workflow }}_${{ env.BRANCH }}_${{ env.DATE }}
+        path: release/*
+        if-no-files-found: error
+        retention-days: 90
 
+    - name: issue评论
+      if: github.event_name != 'pull_request' && github.ref != 'refs/heads/feature/test'
+      uses: actions/github-script@v7
+      with:
+        github-token: ${{ secrets.GITHUB_TOKEN }}
+        script: |
+          github.rest.issues.createComment({
+            issue_number: 483,
+            owner: context.repo.owner,
+            repo: context.repo.repo,
+            body: '- 下载地址: [${{ github.workflow }}_${{ env.BRANCH }}_${{ env.DATE }}](${{ steps.upload.outputs.artifact-url }})\n'
+              + '- 分支: ${{ env.BRANCH2 }}\n' 
+              + '- git hash: ${{ github.sha }} \n' 
+              + '- 编译日期: ${{ env.DATE }}\n' 
+              + '- 打包ci名: ${{ github.workflow }}\n' 
+              + '- 说明: 此二进制为arm64版本\n' 
+          })

.github/workflows/style.yml

@@ -4,7 +4,7 @@ on: [pull_request]
 
 jobs:
   check:
-    runs-on: ubuntu-latest
+    runs-on: ubuntu-20.04
     steps:
       - uses: actions/checkout@v2
         with:

.github/workflows/windows.yml

@@ -17,14 +17,50 @@ jobs:
         with:
           vcpkgDirectory: '${{github.workspace}}/vcpkg'
           vcpkgTriplet: x64-windows-static
-          # 2021.05.12
+          # 2024.06.01
-          vcpkgGitCommitId: '5568f110b509a9fd90711978a7cb76bae75bb092'
+          vcpkgGitCommitId: '47364fbc300756f64f7876b549d9422d5f3ec0d3'
-          vcpkgArguments: 'openssl libsrtp'
+          vcpkgArguments: 'openssl libsrtp[openssl]'
 
       - name: 编译
         uses: lukka/run-cmake@v3
         with:
           useVcpkgToolchainFile: true
           buildDirectory: '${{github.workspace}}/build'
-          cmakeAppendedArgs: '-DCMAKE_ENABLE_WEBRTC:BOOL=ON'
+          cmakeAppendedArgs: ''
-          cmakeBuildType: 'RelWithDebInfo'
+          cmakeBuildType: 'Release'
+
+      - name: 设置环境变量
+        run: |
+          $dateString = Get-Date -Format "yyyy-MM-dd"
+          $branch = $env:GITHUB_REF -replace "refs/heads/", "" -replace "[\\/\\\?\%\*:\|\x22<>]", "_"
+          $branch2 = $env:GITHUB_REF -replace "refs/heads/", ""
+          echo "BRANCH=$branch" >> $env:GITHUB_ENV
+          echo "BRANCH2=$branch2" >> $env:GITHUB_ENV
+          echo "DATE=$dateString" >> $env:GITHUB_ENV
+
+      - name: 打包二进制
+        id: upload
+        uses: actions/upload-artifact@v4
+        with:
+          name: ${{ github.workflow }}_${{ env.BRANCH }}_${{ env.DATE }}
+          path: release/*
+          if-no-files-found: error
+          retention-days: 90
+
+      - name: issue评论
+        if: github.event_name != 'pull_request' && github.ref != 'refs/heads/feature/test'
+        uses: actions/github-script@v7
+        with:
+          github-token: ${{ secrets.GITHUB_TOKEN }}
+          script: |
+            github.rest.issues.createComment({
+              issue_number: 483,
+              owner: context.repo.owner,
+              repo: context.repo.repo,
+              body: '- 下载地址: [${{ github.workflow }}_${{ env.BRANCH }}_${{ env.DATE }}](${{ steps.upload.outputs.artifact-url }})\n'
+                + '- 分支: ${{ env.BRANCH2 }}\n' 
+                + '- git hash: ${{ github.sha }} \n' 
+                + '- 编译日期: ${{ env.DATE }}\n' 
+                + '- 打包ci名: ${{ github.workflow }}\n' 
+                + '- 说明: 此二进制为x64版本\n' 
+            })

3rdpart/ZLToolKit

@@ -1 +1 @@
-Subproject commit 26d54bbc7b1860a450434dce49bbc8fcbcbae88b
+Subproject commit 5144e2aa521df6d473308bfb31172054772a634f

3rdpart/assert.h

@@ -13,22 +13,18 @@
 
 #include <stdio.h>
 
-#ifndef NDEBUG
+#ifdef assert
-    #ifdef assert
     #undef assert
-    #endif//assert
+#endif//assert
 
-    #ifdef __cplusplus
+#ifdef __cplusplus
-    extern "C" {
+extern "C" {
-    #endif
+#endif
-    extern void Assert_Throw(int failed, const char *exp, const char *func, const char *file, int line, const char *str);
+extern void Assert_Throw(int failed, const char *exp, const char *func, const char *file, int line, const char *str);
-    #ifdef __cplusplus
+#ifdef __cplusplus
-    }
+}
-    #endif
+#endif
 
-    #define assert(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__, NULL)
+#define assert(exp) Assert_Throw(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__, NULL)
-#else
-    #define	assert(e)	((void)0)
-#endif//NDEBUG
 
 #endif //ZLMEDIAKIT_ASSERT_H

CMakeLists.txt

@@ -403,6 +403,8 @@ if(OPENSSL_FOUND AND ENABLE_OPENSSL)
   update_cached_list(MK_LINK_LIBRARIES ${OPENSSL_LIBRARIES})
   if(CMAKE_SYSTEM_NAME MATCHES "Linux" AND OPENSSL_USE_STATIC_LIBS)
     update_cached_list(MK_LINK_LIBRARIES ${CMAKE_DL_LIBS})
+  elseif(CMAKE_SYSTEM_NAME MATCHES "Windows" AND OPENSSL_USE_STATIC_LIBS)
+    update_cached_list(MK_LINK_LIBRARIES Crypt32)
   endif()
 else()
   set(ENABLE_OPENSSL OFF)

README.md

@@ -159,12 +159,15 @@
  - 2、作为独立的流媒体服务器使用，不想做c/c++开发的，可以参考 [restful api](https://github.com/ZLMediaKit/ZLMediaKit/wiki/MediaServer支持的HTTP-API) 和 [web hook](https://github.com/ZLMediaKit/ZLMediaKit/wiki/MediaServer支持的HTTP-HOOK-API ).
  - 3、如果想做c/c++开发，添加业务逻辑增加功能，可以参考这里的[测试程序](https://github.com/ZLMediaKit/ZLMediaKit/tree/master/tests).
 
+## 二进制文件下载
+zlmediakit采用 github action 持续集成自动编译打包上传编译产出包，请在[issue列表](https://github.com/ZLMediaKit/ZLMediaKit/issues/483)下载最新sdk库文件以及可执行文件。
+
 ## Docker 镜像
 
 你可以从Docker Hub下载已经编译好的镜像并启动它：
 
 ```bash
-#此镜像为github持续集成自动编译推送，跟代码(master分支)保持最新状态
+#此镜像为github action 持续集成自动编译推送，跟代码(master分支)保持最新状态
 docker run -id -p 1935:1935 -p 8080:80 -p 8443:443 -p 8554:554 -p 10000:10000 -p 10000:10000/udp -p 8000:8000/udp -p 9000:9000/udp zlmediakit/zlmediakit:master
 ```
 
@@ -188,6 +191,7 @@ bash build_docker_images.sh
    - [jessibuca](https://github.com/langhuihui/jessibuca) 基于wasm支持H265的播放器
    - [wsPlayer](https://github.com/v354412101/wsPlayer) 基于MSE的websocket-fmp4播放器
    - [BXC_gb28181Player](https://github.com/any12345com/BXC_gb28181Player) C++开发的支持国标GB28181协议的视频流播放器
+   - [RTCPlayer](https://github.com/leo94666/RTCPlayer) 一个基于Android客户端的的RTC播放器
 
 - WEB管理网站
    - [zlm_webassist](https://github.com/1002victor/zlm_webassist) 本项目配套的前后端分离web管理项目
@@ -222,7 +226,7 @@ bash build_docker_images.sh
  - 请关注微信公众号获取最新消息推送：
  <img src=https://user-images.githubusercontent.com/11495632/232451702-4c50bc72-84d8-4c94-af2b-57290088ba7a.png width=15% />
  
- - 也可以自愿有偿加入知识星球咨询和获取资料：
+ - 也可以自愿有偿加入知识星球咨询、获取资料以及加入微信技术群：
  <img src= https://user-images.githubusercontent.com/11495632/231946329-aa8517b0-3cf5-49cf-8c75-a93ed58cb9d2.png width=30% />
   
 
@@ -363,6 +367,7 @@ bash build_docker_images.sh
 [jamesZHANG500](https://github.com/jamesZHANG500)
 [weidelong](https://github.com/wdl1697454803)
 [小强先生](https://github.com/linshangqiang)
+[李之阳](https://github.com/leo94666)
 
 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

@@ -324,6 +324,10 @@ git submodule update --init
 	});
 	
 	```
+
+## Binary file download
+zlmediakit uses github action to continuously integrate automatic compilation package and upload the compilation output package. Please download the latest sdk library file and executable file at [issue list] (https://github.com/ZLMediaKit/ZLMediaKit/issues/483).
+
 ## Docker Image
 
 You can download the pre-compiled image from Docker Hub and start it:
@@ -369,6 +373,8 @@ bash build_docker_images.sh
   - [WebSocket-fmp4 player based on MSE](https://github.com/v354412101/wsPlayer) 
   - [Domestic webrtc sdk(metaRTC)](https://github.com/metartc/metaRTC)
   - [GB28181 player implemented in C++](https://github.com/any12345com/BXC_gb28181Player)
+  - [Android RTCPlayer](https://github.com/leo94666/RTCPlayer)
+
 
 ## License
 
@@ -521,6 +527,7 @@ Thanks to all those who have supported this project in various ways, including b
 [jamesZHANG500](https://github.com/jamesZHANG500)
 [weidelong](https://github.com/wdl1697454803)
 [小强先生](https://github.com/linshangqiang)
+[李之阳](https://github.com/leo94666)
 
 Also thank to JetBrains for their support for open source project, we developed and debugged zlmediakit with CLion:
 

api/include/mk_common.h

@@ -51,6 +51,9 @@ extern "C" {
 //输出日志到回调函数(mk_events::on_mk_log)
 #define LOG_CALLBACK    (1 << 2)
 
+//向下兼容
+#define mk_env_init1 mk_env_init2
+
 //回调user_data回调函数
 typedef void(API_CALL *on_user_data_free)(void *user_data);
 
@@ -104,7 +107,7 @@ API_EXPORT void API_CALL mk_stop_all_server();
  * @param ssl ssl证书内容或路径，可以为NULL
  * @param ssl_pwd 证书密码，可以为NULL
  */
-API_EXPORT void API_CALL mk_env_init1(int thread_num,
+API_EXPORT void API_CALL mk_env_init2(int thread_num,
                                       int log_level,
                                       int log_mask,
                                       const char *log_file_path,

api/include/mk_events.h

@@ -132,7 +132,12 @@ typedef struct {
     /**
      * 录制mp4分片文件成功后广播
      */
-    void (API_CALL *on_mk_record_mp4)(const mk_mp4_info mp4);
+    void (API_CALL *on_mk_record_mp4)(const mk_record_info mp4);
+
+     /**
+     * 录制ts分片文件成功后广播
+     */
+    void (API_CALL *on_mk_record_ts)(const mk_record_info ts);
 
     /**
      * shell登录鉴权
@@ -175,16 +180,16 @@ typedef struct {
      * @param err 错误代码
      * @param msg 错误提示
      */
-    void(API_CALL *on_mk_media_send_rtp_stop)(const char *vhost, const char *app, const char *stream, const char *ssrc, int err, const char *msg);
+    void (API_CALL *on_mk_media_send_rtp_stop)(const char *vhost, const char *app, const char *stream, const char *ssrc, int err, const char *msg);
 
     /**
      * rtc sctp连接中/完成/失败/关闭回调
      * @param rtc_transport 数据通道对象
      */
-    void(API_CALL *on_mk_rtc_sctp_connecting)(mk_rtc_transport rtc_transport);
+    void (API_CALL *on_mk_rtc_sctp_connecting)(mk_rtc_transport rtc_transport);
-    void(API_CALL *on_mk_rtc_sctp_connected)(mk_rtc_transport rtc_transport);
+    void (API_CALL *on_mk_rtc_sctp_connected)(mk_rtc_transport rtc_transport);
-    void(API_CALL *on_mk_rtc_sctp_failed)(mk_rtc_transport rtc_transport);
+    void (API_CALL *on_mk_rtc_sctp_failed)(mk_rtc_transport rtc_transport);
-    void(API_CALL *on_mk_rtc_sctp_closed)(mk_rtc_transport rtc_transport);
+    void (API_CALL *on_mk_rtc_sctp_closed)(mk_rtc_transport rtc_transport);
 
     /**
      * rtc数据通道发送数据回调
@@ -192,7 +197,7 @@ typedef struct {
      * @param msg 数据
      * @param len 数据长度
      */
-    void(API_CALL *on_mk_rtc_sctp_send)(mk_rtc_transport rtc_transport, const uint8_t *msg, size_t len);
+    void (API_CALL *on_mk_rtc_sctp_send)(mk_rtc_transport rtc_transport, const uint8_t *msg, size_t len);
 
     /**
      * rtc数据通道接收数据回调
@@ -202,7 +207,7 @@ typedef struct {
      * @param msg 数据
      * @param len 数据长度
      */
-    void(API_CALL *on_mk_rtc_sctp_received)(mk_rtc_transport rtc_transport, uint16_t streamId, uint32_t ppid, const uint8_t *msg, size_t len);
+    void (API_CALL *on_mk_rtc_sctp_received)(mk_rtc_transport rtc_transport, uint16_t streamId, uint32_t ppid, const uint8_t *msg, size_t len);
 
 } mk_events;
 

api/include/mk_events_objects.h

@@ -18,29 +18,42 @@
 extern "C" {
 #endif
 
-///////////////////////////////////////////MP4Info/////////////////////////////////////////////
+///////////////////////////////////////////RecordInfo/////////////////////////////////////////////
-//MP4Info对象的C映射
+//RecordInfo对象的C映射
-typedef struct mk_mp4_info_t *mk_mp4_info;
+typedef struct mk_record_info_t *mk_record_info;
 // GMT 标准时间，单位秒
-API_EXPORT uint64_t API_CALL mk_mp4_info_get_start_time(const mk_mp4_info ctx);
+API_EXPORT uint64_t API_CALL mk_record_info_get_start_time(const mk_record_info ctx);
 // 录像长度，单位秒
-API_EXPORT float API_CALL mk_mp4_info_get_time_len(const mk_mp4_info ctx);
+API_EXPORT float API_CALL mk_record_info_get_time_len(const mk_record_info ctx);
 // 文件大小，单位 BYTE
-API_EXPORT size_t API_CALL mk_mp4_info_get_file_size(const mk_mp4_info ctx);
+API_EXPORT size_t API_CALL mk_record_info_get_file_size(const mk_record_info ctx);
 // 文件路径
-API_EXPORT const char* API_CALL mk_mp4_info_get_file_path(const mk_mp4_info ctx);
+API_EXPORT const char *API_CALL mk_record_info_get_file_path(const mk_record_info ctx);
 // 文件名称
-API_EXPORT const char* API_CALL mk_mp4_info_get_file_name(const mk_mp4_info ctx);
+API_EXPORT const char *API_CALL mk_record_info_get_file_name(const mk_record_info ctx);
 // 文件夹路径
-API_EXPORT const char* API_CALL mk_mp4_info_get_folder(const mk_mp4_info ctx);
+API_EXPORT const char *API_CALL mk_record_info_get_folder(const mk_record_info ctx);
 // 播放路径
-API_EXPORT const char* API_CALL mk_mp4_info_get_url(const mk_mp4_info ctx);
+API_EXPORT const char *API_CALL mk_record_info_get_url(const mk_record_info ctx);
 // 应用名称
-API_EXPORT const char* API_CALL mk_mp4_info_get_vhost(const mk_mp4_info ctx);
+API_EXPORT const char *API_CALL mk_record_info_get_vhost(const mk_record_info ctx);
 // 流 ID
-API_EXPORT const char* API_CALL mk_mp4_info_get_app(const mk_mp4_info ctx);
+API_EXPORT const char *API_CALL mk_record_info_get_app(const mk_record_info ctx);
 // 虚拟主机
-API_EXPORT const char* API_CALL mk_mp4_info_get_stream(const mk_mp4_info ctx);
+API_EXPORT const char *API_CALL mk_record_info_get_stream(const mk_record_info ctx);
+
+//// 下面宏保障用户代码兼容性, 二进制abi不兼容，用户需要重新编译链接 /////
+#define mk_mp4_info mk_record_info
+#define mk_mp4_info_get_start_time mk_record_info_get_start_time
+#define mk_mp4_info_get_time_len mk_record_info_get_time_len
+#define mk_mp4_info_get_file_size mk_record_info_get_file_size
+#define mk_mp4_info_get_file_path mk_record_info_get_file_path
+#define mk_mp4_info_get_file_name mk_record_info_get_file_name
+#define mk_mp4_info_get_folder mk_record_info_get_folder
+#define mk_mp4_info_get_url mk_record_info_get_url
+#define mk_mp4_info_get_vhost mk_record_info_get_vhost
+#define mk_mp4_info_get_app mk_record_info_get_app
+#define mk_mp4_info_get_stream mk_record_info_get_stream
 
 ///////////////////////////////////////////Parser/////////////////////////////////////////////
 //Parser对象的C映射

api/include/mk_proxyplayer.h

@@ -32,6 +32,7 @@ typedef struct mk_proxy_player_t *mk_proxy_player;
  */
 API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create(const char *vhost, const char *app, const char *stream, int hls_enabled, int mp4_enabled);
 
+
 /**
  * 创建一个代理播放器
  * @param vhost 虚拟主机名，一般为__defaultVhost__
@@ -43,6 +44,32 @@ API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create(const char *vhost, co
 API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create2(const char *vhost, const char *app, const char *stream, mk_ini option);
 
 
+/**
+ * 创建一个代理播放器
+ * @param vhost 虚拟主机名，一般为__defaultVhost__
+ * @param app 应用名
+ * @param stream 流名
+ * @param rtp_type rtsp播放方式:RTP_TCP = 0, RTP_UDP = 1, RTP_MULTICAST = 2
+ * @param hls_enabled 是否生成hls
+ * @param mp4_enabled 是否生成mp4
+ * @param retry_count 重试次数，当<0无限次重试
+ * @return 对象指针
+ */
+API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create3(const char *vhost, const char *app, const char *stream, int hls_enabled, int mp4_enabled, int retry_count);
+
+
+/**
+ * 创建一个代理播放器
+ * @param vhost 虚拟主机名，一般为__defaultVhost__
+ * @param app 应用名
+ * @param stream 流名
+ * @param option ProtocolOption相关配置
+ * @param retry_count 重试次数，当<0无限次重试
+ * @return 对象指针
+ */
+API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create4(const char *vhost, const char *app, const char *stream, mk_ini option, int retry_count);
+
+
 /**
  * 销毁代理播放器
  * @param ctx 对象指针
@@ -71,7 +98,9 @@ API_EXPORT void API_CALL mk_proxy_player_play(mk_proxy_player ctx, const char *u
  * 如果你不调用mk_proxy_player_release函数，那么MediaSource.close()操作将无效
  * @param user_data 用户数据指针，通过mk_proxy_player_set_on_close函数设置
  */
-typedef void(API_CALL *on_mk_proxy_player_close)(void *user_data, int err, const char *what, int sys_err);
+typedef void(API_CALL *on_mk_proxy_player_cb)(void *user_data, int err, const char *what, int sys_err);
+// 保持兼容
+#define on_mk_proxy_player_close on_mk_proxy_player_cb
 
 /**
  * 监听MediaSource.close()事件
@@ -81,8 +110,17 @@ typedef void(API_CALL *on_mk_proxy_player_close)(void *user_data, int err, const
  * @param cb 回调指针
  * @param user_data 用户数据指针
  */
-API_EXPORT void API_CALL mk_proxy_player_set_on_close(mk_proxy_player ctx, on_mk_proxy_player_close cb, void *user_data);
+API_EXPORT void API_CALL mk_proxy_player_set_on_close(mk_proxy_player ctx, on_mk_proxy_player_cb cb, void *user_data);
-API_EXPORT void API_CALL mk_proxy_player_set_on_close2(mk_proxy_player ctx, on_mk_proxy_player_close cb, void *user_data, on_user_data_free user_data_free);
+API_EXPORT void API_CALL mk_proxy_player_set_on_close2(mk_proxy_player ctx, on_mk_proxy_player_cb cb, void *user_data, on_user_data_free user_data_free);
+
+/**
+ * 设置代理第一次播放结果回调，如果第一次播放失败，可以认作是启动失败
+ * @param ctx 对象指针
+ * @param cb 回调指针
+ * @param user_data 用户数据指针
+ * @param user_data_free 用户数据释放回调
+ */
+API_EXPORT void API_CALL mk_proxy_player_set_on_play_result(mk_proxy_player ctx, on_mk_proxy_player_cb cb, void *user_data, on_user_data_free user_data_free);
 
 /**
  * 获取总的观看人数

api/include/mk_util.h

@@ -128,6 +128,15 @@ API_EXPORT char *API_CALL mk_ini_dump_string(mk_ini ini);
  * @param file 配置文件路径
  */
 API_EXPORT void API_CALL mk_ini_dump_file(mk_ini ini, const char *file);
+///////////////////////////////////////////统计/////////////////////////////////////////////
+
+typedef void(API_CALL *on_mk_get_statistic_cb)(void *user_data, mk_ini ini);
+
+/**
+ * 获取内存数据统计
+ * @param ini 存放统计结果
+ */
+API_EXPORT void API_CALL mk_get_statistic(on_mk_get_statistic_cb cb, void *user_data, on_user_data_free free_cb);
 
 ///////////////////////////////////////////日志/////////////////////////////////////////////
 

api/source/mk_common.cpp

@@ -83,7 +83,7 @@ API_EXPORT void API_CALL mk_stop_all_server(){
     stopAllTcpServer();
 }
 
-API_EXPORT void API_CALL mk_env_init1(int thread_num,
+API_EXPORT void API_CALL mk_env_init2(int thread_num,
                                       int log_level,
                                       int log_mask,
                                       const char *log_file_path,

api/source/mk_events.cpp

@@ -42,7 +42,13 @@ API_EXPORT void API_CALL mk_events_listen(const mk_events *events){
 
         NoticeCenter::Instance().addListener(&s_tag,Broadcast::kBroadcastRecordMP4,[](BroadcastRecordMP4Args){
             if(s_events.on_mk_record_mp4){
-                s_events.on_mk_record_mp4((mk_mp4_info)&info);
+                s_events.on_mk_record_mp4((mk_record_info)&info);
+            }
+        });
+
+        NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRecordTs, [](BroadcastRecordTsArgs) {
+            if (s_events.on_mk_record_ts) {
+                s_events.on_mk_record_ts((mk_record_info)&info);
             }
         });
 

api/source/mk_events_objects.cpp

@@ -26,61 +26,61 @@ using namespace toolkit;
 using namespace mediakit;
 
 ///////////////////////////////////////////RecordInfo/////////////////////////////////////////////
-API_EXPORT uint64_t API_CALL mk_mp4_info_get_start_time(const mk_mp4_info ctx){
+API_EXPORT uint64_t API_CALL mk_record_info_get_start_time(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->start_time;
 }
 
-API_EXPORT float API_CALL mk_mp4_info_get_time_len(const mk_mp4_info ctx){
+API_EXPORT float API_CALL mk_record_info_get_time_len(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->time_len;
 }
 
-API_EXPORT size_t API_CALL mk_mp4_info_get_file_size(const mk_mp4_info ctx){
+API_EXPORT size_t API_CALL mk_record_info_get_file_size(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->file_size;
 }
 
-API_EXPORT const char* API_CALL mk_mp4_info_get_file_path(const mk_mp4_info ctx){
+API_EXPORT const char *API_CALL mk_record_info_get_file_path(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->file_path.c_str();
 }
 
-API_EXPORT const char* API_CALL mk_mp4_info_get_file_name(const mk_mp4_info ctx){
+API_EXPORT const char *API_CALL mk_record_info_get_file_name(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->file_name.c_str();
 }
 
-API_EXPORT const char* API_CALL mk_mp4_info_get_folder(const mk_mp4_info ctx){
+API_EXPORT const char *API_CALL mk_record_info_get_folder(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->folder.c_str();
 }
 
-API_EXPORT const char* API_CALL mk_mp4_info_get_url(const mk_mp4_info ctx){
+API_EXPORT const char *API_CALL mk_record_info_get_url(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->url.c_str();
 }
 
-API_EXPORT const char* API_CALL mk_mp4_info_get_vhost(const mk_mp4_info ctx){
+API_EXPORT const char *API_CALL mk_record_info_get_vhost(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->vhost.c_str();
 }
 
-API_EXPORT const char* API_CALL mk_mp4_info_get_app(const mk_mp4_info ctx){
+API_EXPORT const char *API_CALL mk_record_info_get_app(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->app.c_str();
 }
 
-API_EXPORT const char* API_CALL mk_mp4_info_get_stream(const mk_mp4_info ctx){
+API_EXPORT const char *API_CALL mk_record_info_get_stream(const mk_record_info ctx) {
     assert(ctx);
     RecordInfo *info = (RecordInfo *)ctx;
     return info->stream.c_str();
@@ -528,7 +528,7 @@ API_EXPORT void API_CALL mk_auth_invoker_clone_release(const mk_auth_invoker ctx
 }
 
 ///////////////////////////////////////////WebRtcTransport/////////////////////////////////////////////
-API_EXPORT void API_CALL mk_rtc_sendDatachannel(const mk_rtc_transport ctx, uint16_t streamId, uint32_t ppid, const char *msg, size_t len) {
+API_EXPORT void API_CALL mk_rtc_send_datachannel(const mk_rtc_transport ctx, uint16_t streamId, uint32_t ppid, const char *msg, size_t len) {
 #ifdef ENABLE_WEBRTC
     assert(ctx && msg);
     WebRtcTransport *transport = (WebRtcTransport *)ctx;

api/source/mk_h264_splitter.cpp

@@ -27,8 +27,9 @@ protected:
 
 private:
     bool _h265 = false;
+    bool _have_decode_frame = false;
     onH264 _cb;
-    size_t _search_pos = 0;
+    toolkit::BufferLikeString _buffer;
 };
 
 void H264Splitter::setOnSplitted(H264Splitter::onH264 cb) {
@@ -42,11 +43,21 @@ H264Splitter::~H264Splitter() {
 }
 
 ssize_t H264Splitter::onRecvHeader(const char *data, size_t len) {
-    _cb(data, len);
+    auto frame = Factory::getFrameFromPtr(_h265 ? CodecH265 : CodecH264, (char *)data, len, 0, 0);
+    if (_have_decode_frame && (frame->decodeAble() || frame->configFrame())) {
+        // 缓存中存在可解码帧，且下一帧是可解码帧或者配置帧，那么flush缓存
+        _cb(_buffer.data(), _buffer.size());
+        _buffer.assign(data, len);
+        _have_decode_frame = frame->decodeAble();
+    } else {
+        // 还需要缓存
+        _buffer.append(data, len);
+        _have_decode_frame = _have_decode_frame || frame->decodeAble();
+    }
     return 0;
 }
 
-static const char *onSearchPacketTail_l(const char *data, size_t len) {
+const char *H264Splitter::onSearchPacketTail(const char *data, size_t len) {
     for (size_t i = 2; len > 2 && i < len - 2; ++i) {
         //判断0x00 00 01
         if (data[i] == 0 && data[i + 1] == 0 && data[i + 2] == 1) {
@@ -60,28 +71,6 @@ static const char *onSearchPacketTail_l(const char *data, size_t len) {
     return nullptr;
 }
 
-const char *H264Splitter::onSearchPacketTail(const char *data, size_t len) {
-    auto last_frame = data + _search_pos;
-    auto next_frame = onSearchPacketTail_l(last_frame, len - _search_pos);
-    if (!next_frame) {
-        return nullptr;
-    }
-
-    auto last_frame_len = next_frame - last_frame;
-    Frame::Ptr frame;
-    if (_h265) {
-        frame = Factory::getFrameFromPtr(CodecH265, (char *)last_frame, last_frame_len, 0, 0);
-    } else {
-        frame = Factory::getFrameFromPtr(CodecH264, (char *)last_frame, last_frame_len, 0, 0);
-    }
-    if (frame->decodeAble()) {
-        _search_pos = 0;
-        return next_frame;
-    }
-    _search_pos += last_frame_len;
-    return nullptr;
-}
-
 ////////////////////////////////////////////////////////////////////////////////////////////////////////
 
 API_EXPORT mk_h264_splitter API_CALL mk_h264_splitter_create(on_mk_h264_splitter_frame cb, void *user_data, int is_h265) {

api/source/mk_proxyplayer.cpp

@@ -16,22 +16,30 @@ using namespace toolkit;
 using namespace mediakit;
 
 API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create(const char *vhost, const char *app, const char *stream, int hls_enabled, int mp4_enabled) {
+   return mk_proxy_player_create3(vhost, app, stream, hls_enabled, mp4_enabled,-1);
+}
+
+API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create2(const char *vhost, const char *app, const char *stream, mk_ini ini) {
+    return mk_proxy_player_create4(vhost, app, stream, ini, -1);
+}
+
+API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create3(const char *vhost, const char *app, const char *stream, int hls_enabled, int mp4_enabled, int retry_count) {
     assert(vhost && app && stream);
     ProtocolOption option;
     option.enable_hls = hls_enabled;
     option.enable_mp4 = mp4_enabled;
-    PlayerProxy::Ptr *obj(new PlayerProxy::Ptr(new PlayerProxy(vhost, app, stream, option)));
+    PlayerProxy::Ptr *obj(new PlayerProxy::Ptr(new PlayerProxy(vhost, app, stream, option, retry_count)));
-    return (mk_proxy_player) obj;
-}
-
-API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create2(const char *vhost, const char *app, const char *stream, mk_ini ini) {
-    assert(vhost && app && stream);
-    ProtocolOption option(*((mINI *)ini));
-    PlayerProxy::Ptr *obj(new PlayerProxy::Ptr(new PlayerProxy(vhost, app, stream, option)));
     return (mk_proxy_player)obj;
 }
 
 
+API_EXPORT mk_proxy_player API_CALL mk_proxy_player_create4(const char *vhost, const char *app, const char *stream, mk_ini ini, int retry_count) {
+    assert(vhost && app && stream);
+    ProtocolOption option(*((mINI *)ini));
+    PlayerProxy::Ptr *obj(new PlayerProxy::Ptr(new PlayerProxy(vhost, app, stream, option, retry_count)));
+    return (mk_proxy_player)obj;
+}
+
 API_EXPORT void API_CALL mk_proxy_player_release(mk_proxy_player ctx) {
     assert(ctx);
     PlayerProxy::Ptr *obj = (PlayerProxy::Ptr *) ctx;
@@ -76,6 +84,20 @@ API_EXPORT void API_CALL mk_proxy_player_set_on_close2(mk_proxy_player ctx, on_m
     });
 }
 
+API_EXPORT void API_CALL mk_proxy_player_set_on_play_result(mk_proxy_player ctx, on_mk_proxy_player_close cb, void *user_data, on_user_data_free user_data_free) {
+    assert(ctx);
+    PlayerProxy::Ptr &obj = *((PlayerProxy::Ptr *)ctx);
+    std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
+    obj->getPoller()->async([obj, cb, ptr]() {
+        // 切换线程再操作
+        obj->setPlayCallbackOnce([cb, ptr](const SockException &ex) {
+            if (cb) {
+                cb(ptr.get(), ex.getErrCode(), ex.what(), ex.getCustomCode());
+            }
+        });
+    });
+}
+
 API_EXPORT int API_CALL mk_proxy_player_total_reader_count(mk_proxy_player ctx){
     assert(ctx);
     PlayerProxy::Ptr &obj = *((PlayerProxy::Ptr *) ctx);

api/source/mk_rtp_server.cpp

@@ -56,7 +56,7 @@ API_EXPORT void API_CALL mk_rtp_server_set_on_detach2(mk_rtp_server ctx, on_mk_r
     RtpServer::Ptr *server = (RtpServer::Ptr *) ctx;
     if (cb) {
         std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
-        (*server)->setOnDetach([cb, ptr]() {
+        (*server)->setOnDetach([cb, ptr](const SockException &ex) {
             cb(ptr.get());
         });
     } else {

api/source/mk_util.cpp

@@ -15,6 +15,9 @@
 #include "Util/util.h"
 #include "Util/mini.h"
 #include "Util/logger.h"
+#include "Util/TimeTicker.h"
+#include "Poller/EventPoller.h"
+#include "Thread/WorkThreadPool.h"
 #include "Common/config.h"
 
 using namespace std;
@@ -132,6 +135,129 @@ API_EXPORT void API_CALL mk_ini_dump_file(mk_ini ini, const char *file) {
     ptr->dumpFile(file);
 }
 
+extern uint64_t getTotalMemUsage();
+extern uint64_t getTotalMemBlock();
+extern uint64_t getThisThreadMemUsage();
+extern uint64_t getThisThreadMemBlock();
+extern std::vector<size_t> getBlockTypeSize();
+extern uint64_t getTotalMemBlockByType(int type);
+extern uint64_t getThisThreadMemBlockByType(int type);
+
+namespace mediakit {
+class MediaSource;
+class MultiMediaSourceMuxer;
+class FrameImp;
+class Frame;
+class RtpPacket;
+class RtmpPacket;
+} // namespace mediakit
+
+namespace toolkit {
+class TcpServer;
+class TcpSession;
+class UdpServer;
+class UdpSession;
+class TcpClient;
+class Socket;
+class Buffer;
+class BufferRaw;
+class BufferLikeString;
+class BufferList;
+} // namespace toolkit
+
+API_EXPORT void API_CALL mk_get_statistic(on_mk_get_statistic_cb func, void *user_data, on_user_data_free free_cb) {
+    assert(func);
+    std::shared_ptr<void> data(user_data, free_cb);
+    auto cb = [func, data](const toolkit::mINI &ini) { func(data.get(), (mk_ini)&ini); };
+    auto obj = std::make_shared<toolkit::mINI>();
+    auto &val = *obj;
+
+    val["object.MediaSource"] = ObjectStatistic<MediaSource>::count();
+    val["object.MultiMediaSourceMuxer"] = ObjectStatistic<MultiMediaSourceMuxer>::count();
+
+    val["object.TcpServer"] = ObjectStatistic<TcpServer>::count();
+    val["object.TcpSession"] = ObjectStatistic<TcpSession>::count();
+    val["object.UdpServer"] = ObjectStatistic<UdpServer>::count();
+    val["object.UdpSession"] = ObjectStatistic<UdpSession>::count();
+    val["object.TcpClient"] = ObjectStatistic<TcpClient>::count();
+    val["object.Socket"] = ObjectStatistic<Socket>::count();
+
+    val["object.FrameImp"] = ObjectStatistic<FrameImp>::count();
+    val["object.Frame"] = ObjectStatistic<Frame>::count();
+
+    val["object.Buffer"] = ObjectStatistic<Buffer>::count();
+    val["object.BufferRaw"] = ObjectStatistic<BufferRaw>::count();
+    val["object.BufferLikeString"] = ObjectStatistic<BufferLikeString>::count();
+    val["object.BufferList"] = ObjectStatistic<BufferList>::count();
+
+    val["object.RtpPacket"] = ObjectStatistic<RtpPacket>::count();
+    val["object.RtmpPacket"] = ObjectStatistic<RtmpPacket>::count();
+#ifdef ENABLE_MEM_DEBUG
+    auto bytes = getTotalMemUsage();
+    val["memory.memUsage"] = bytes;
+    val["memory.memUsageMB"] = (int)(bytes / 1024 / 1024);
+    val["memory.memBlock"] = getTotalMemBlock();
+    static auto block_type_size = getBlockTypeSize();
+    {
+        int i = 0;
+        string str;
+        size_t last = 0;
+        for (auto sz : block_type_size) {
+            str.append(to_string(last) + "~" + to_string(sz) + ":" + to_string(getTotalMemBlockByType(i++)) + ";");
+            last = sz;
+        }
+        str.pop_back();
+        val["memory.memBlockTypeCount"] = str;
+    }
+#endif
+
+    auto thread_size = EventPollerPool::Instance().getExecutorSize() + WorkThreadPool::Instance().getExecutorSize();
+    std::shared_ptr<vector<toolkit::mINI>> thread_mem_info = std::make_shared<vector<toolkit::mINI>>(thread_size);
+
+    shared_ptr<void> finished(nullptr, [thread_mem_info, cb, obj](void *) {
+        for (auto &val : *thread_mem_info) {
+            auto thread_name = val["name"];
+            replace(thread_name, "...", "~~~");
+            auto prefix = "thread-" + thread_name + ".";
+            for (auto &pr : val) {
+                (*obj).emplace(prefix + pr.first, std::move(pr.second));
+            }
+        }
+        // 触发回调
+        cb(*obj);
+    });
+
+    auto pos = 0;
+    auto lambda = [&](const TaskExecutor::Ptr &executor) {
+        auto &val = (*thread_mem_info)[pos++];
+        val["load"] = executor->load();
+        Ticker ticker;
+        executor->async([finished, &val, ticker]() {
+            val["name"] = getThreadName();
+            val["delay"] = ticker.elapsedTime();
+#ifdef ENABLE_MEM_DEBUG
+            auto bytes = getThisThreadMemUsage();
+            val["memUsage"] = bytes;
+            val["memUsageMB"] = bytes / 1024 / 1024;
+            val["memBlock"] = getThisThreadMemBlock();
+            {
+                int i = 0;
+                string str;
+                size_t last = 0;
+                for (auto sz : block_type_size) {
+                    str.append(to_string(last) + "~" + to_string(sz) + ":" + to_string(getThisThreadMemBlockByType(i++)) + ";");
+                    last = sz;
+                }
+                str.pop_back();
+                val["memBlockTypeCount"] = str;
+            }
+#endif
+        });
+    };
+    EventPollerPool::Instance().for_each(lambda);
+    WorkThreadPool::Instance().for_each(lambda);
+}
+
 API_EXPORT void API_CALL mk_log_printf(int level, const char *file, const char *function, int line, const char *fmt, ...) {
     va_list ap;
     va_start(ap, fmt);

api/tests/server.c

@@ -189,6 +189,14 @@ static void on_mk_webrtc_get_answer_sdp_func(void *user_data, const char *answer
         free((void *)answer);
     }
 }
+
+void API_CALL on_get_statistic_cb(void *user_data, mk_ini ini) {
+    const char *response_header[] = { NULL };
+    char *str = mk_ini_dump_string(ini);
+    mk_http_response_invoker_do_string(user_data, 200, response_header, str);
+    mk_free(str);
+}
+
 /**
  * 收到http api请求广播(包括GET/POST)
  * @param parser http请求内容对象
@@ -247,6 +255,9 @@ void API_CALL on_mk_http_request(const mk_parser parser,
 
         mk_webrtc_get_answer_sdp(mk_http_response_invoker_clone(invoker), on_mk_webrtc_get_answer_sdp_func,
                                  mk_parser_get_url_param(parser, "type"), mk_parser_get_content(parser, NULL), rtc_url);
+    } else if (strcmp(url, "/index/api/getStatistic") == 0) {
+        //拦截api: /index/api/webrtc
+        mk_get_statistic(on_get_statistic_cb, mk_http_response_invoker_clone(invoker), (on_user_data_free) mk_http_response_invoker_clone_release);
     } else {
         *consumed = 0;
         return;
@@ -387,7 +398,7 @@ void API_CALL on_mk_rtsp_auth(const mk_media_info url_info,
 /**
  * 录制mp4分片文件成功后广播
  */
-void API_CALL on_mk_record_mp4(const mk_mp4_info mp4) {
+void API_CALL on_mk_record_mp4(const mk_record_info mp4) {
     log_printf(LOG_LEV,
                "\nstart_time: %d\n"
                "time_len: %d\n"
@@ -399,16 +410,16 @@ void API_CALL on_mk_record_mp4(const mk_mp4_info mp4) {
                "vhost: %s\n"
                "app: %s\n"
                "stream: %s\n",
-               mk_mp4_info_get_start_time(mp4),
+               mk_record_info_get_start_time(mp4),
-               mk_mp4_info_get_time_len(mp4),
+               mk_record_info_get_time_len(mp4),
-               mk_mp4_info_get_file_size(mp4),
+               mk_record_info_get_file_size(mp4),
-               mk_mp4_info_get_file_path(mp4),
+               mk_record_info_get_file_path(mp4),
-               mk_mp4_info_get_file_name(mp4),
+               mk_record_info_get_file_name(mp4),
-               mk_mp4_info_get_folder(mp4),
+               mk_record_info_get_folder(mp4),
-               mk_mp4_info_get_url(mp4),
+               mk_record_info_get_url(mp4),
-               mk_mp4_info_get_vhost(mp4),
+               mk_record_info_get_vhost(mp4),
-               mk_mp4_info_get_app(mp4),
+               mk_record_info_get_app(mp4),
-               mk_mp4_info_get_stream(mp4));
+               mk_record_info_get_stream(mp4));
 }
 
 /**

conf/config.ini

@@ -126,6 +126,8 @@ wait_track_ready_ms=10000
 wait_add_track_ms=3000
 #如果track未就绪，我们先缓存帧数据，但是有最大个数限制，防止内存溢出
 unready_frame_cache=100
+#是否启用观看人数变化事件广播，置1则启用，置0则关闭
+broadcast_player_count_changed=0
 
 [hls]
 #hls写文件的buf大小，调整参数可以提高文件io性能
@@ -367,6 +369,24 @@ start_bitrate=0
 max_bitrate=0
 min_bitrate=0
 
+#nack接收端
+#Nack缓存包最早时间间隔
+maxNackMS=5000
+#Nack包检查间隔(包数量)
+rtpCacheCheckInterval=100
+
+#nack发送端
+#最大保留的rtp丢包状态个数
+nackMaxSize=2048
+#rtp丢包状态最长保留时间
+nackMaxMS=3000
+#nack最多请求重传次数
+nackMaxCount=15
+#nack重传频率，rtt的倍数
+nackIntervalRatio=1.0
+#nack包中rtp个数，减小此值可以让nack包响应更灵敏
+nackRtpSize=8
+
 [srt]
 #srt播放推流、播放超时时间,单位秒
 timeoutSec=5

dockerfile

@@ -43,7 +43,7 @@ WORKDIR /opt/media/ZLMediaKit
 
 # 3rdpart init
 WORKDIR /opt/media/ZLMediaKit/3rdpart
-RUN wget https://mirror.ghproxy.com/https://github.com/cisco/libsrtp/archive/v2.3.0.tar.gz -O libsrtp-2.3.0.tar.gz && \
+RUN wget https://github.com/cisco/libsrtp/archive/v2.3.0.tar.gz -O libsrtp-2.3.0.tar.gz && \
     tar xfv libsrtp-2.3.0.tar.gz && \
     mv libsrtp-2.3.0 libsrtp && \
     cd libsrtp && ./configure --enable-openssl && make -j $(nproc) && make install

ext-codec/SPSParser.c

@@ -306,13 +306,12 @@ static int getBits(void *pvHandle, int iN)
     uint8_t u8Nbyte;
     uint8_t u8Shift;
     uint32_t u32Result = 0;
-    int iRet = 0;
+    uint32_t iRet = 0;
     int iResoLen = 0;
 
     if(NULL == ptPtr)
     {
         RPT(RPT_ERR, "NULL pointer");
-        iRet = -1;
         goto exit;
     }
 
@@ -324,7 +323,6 @@ static int getBits(void *pvHandle, int iN)
     iResoLen = getBitsLeft(ptPtr);
     if(iResoLen < iN)
     {
-        iRet = -1;
         goto exit;
     }
 

server/System.cpp

@@ -22,11 +22,12 @@
 #include <map>
 #include <iostream>
 
-#include "Common/JemallocUtil.h"
-#include "Common/macros.h"
 #include "System.h"
+#include "Util/util.h"
 #include "Util/logger.h"
 #include "Util/uv_errno.h"
+#include "Common/macros.h"
+#include "Common/JemallocUtil.h"
 
 using namespace std;
 using namespace toolkit;
@@ -66,6 +67,16 @@ static void save_jemalloc_stats() {
     out.flush();
 }
 
+static std::string get_func_symbol(const std::string &symbol) {
+    size_t pos1 = symbol.find("(");
+    if (pos1 == string::npos) {
+        return "";
+    }
+    size_t pos2 = symbol.find("+", pos1);
+    auto ret = symbol.substr(pos1 + 1, pos2 - pos1 - 1);
+    return ret;
+}
+
 static void sig_crash(int sig) {
     signal(sig, SIG_DFL);
     void *array[MAX_STACK_FRAMES];
@@ -78,6 +89,10 @@ static void sig_crash(int sig) {
         std::string symbol(strings[i]);
         ref.emplace_back(symbol);
 #if defined(__linux) || defined(__linux__)
+        auto func_symbol = get_func_symbol(symbol);
+        if (!func_symbol.empty()) {
+            ref.emplace_back(toolkit::demangle(func_symbol.data()));
+        }
         static auto addr2line = [](const string &address) {
             string cmd = StrPrinter << "addr2line -C -f -e " << exePath() << " " << address;
             return System::execute(cmd);

server/WebApi.cpp

@@ -45,7 +45,7 @@
 #include "Http/HttpRequester.h"
 #include "Player/PlayerProxy.h"
 #include "Pusher/PusherProxy.h"
-#include "Rtp/RtpSelector.h"
+#include "Rtp/RtpProcess.h"
 #include "Record/MP4Reader.h"
 
 #if defined(ENABLE_RTPPROXY)
@@ -289,6 +289,8 @@ static inline void addHttpListener(){
             it->second(parser, invoker, sender);
         } catch (ApiRetException &ex) {
             responseApi(ex.code(), ex.what(), invoker);
+            auto helper = static_cast<SocketHelper &>(sender).shared_from_this();
+            helper->getPoller()->async([helper, ex]() { helper->shutdown(SockException(Err_shutdown, ex.what())); }, false);
         }
 #ifdef ENABLE_MYSQL
         catch(SqlException &ex){
@@ -321,6 +323,11 @@ public:
         return _map.erase(key);
     }
 
+    size_t size() { 
+        std::lock_guard<std::recursive_mutex> lck(_mtx);
+        return _map.size();
+    }
+
     Pointer find(const std::string &key) const {
         std::lock_guard<std::recursive_mutex> lck(_mtx);
         auto it = _map.find(key);
@@ -478,7 +485,7 @@ uint16_t openRtpServer(uint16_t local_port, const string &stream_id, int tcp_mod
     auto server = s_rtp_server.makeWithAction(stream_id, [&](RtpServer::Ptr server) {
         server->start(local_port, stream_id, (RtpServer::TcpMode)tcp_mode, local_ip.c_str(), re_use_port, ssrc, only_track, multiplex);
     });
-    server->setOnDetach([stream_id]() {
+    server->setOnDetach([stream_id](const SockException &ex) {
         //设置rtp超时移除事件
         s_rtp_server.erase(stream_id);
     });
@@ -1191,8 +1198,8 @@ void installWebApi() {
     api_regist("/index/api/getRtpInfo",[](API_ARGS_MAP){
         CHECK_SECRET();
         CHECK_ARGS("stream_id");
-
+        auto src = MediaSource::find(DEFAULT_VHOST, kRtpAppName, allArgs["stream_id"]);
-        auto process = RtpSelector::Instance().getProcess(allArgs["stream_id"], false);
+        auto process = src ? src->getRtpProcess() : nullptr;
         if (!process) {
             val["exist"] = false;
             return;
@@ -1431,9 +1438,10 @@ void installWebApi() {
         CHECK_SECRET();
         CHECK_ARGS("stream_id");
         //只是暂停流的检查，流媒体服务器做为流负载服务，收流就转发，RTSP/RTMP有自己暂停协议
-        auto rtp_process = RtpSelector::Instance().getProcess(allArgs["stream_id"], false);
+        auto src = MediaSource::find(DEFAULT_VHOST, kRtpAppName, allArgs["stream_id"]);
-        if (rtp_process) {
+        auto process = src ? src->getRtpProcess() : nullptr;
-            rtp_process->setStopCheckRtp(true);
+        if (process) {
+            process->setStopCheckRtp(true);
         } else {
             val["code"] = API::NotFound;
         }
@@ -1442,9 +1450,10 @@ void installWebApi() {
     api_regist("/index/api/resumeRtpCheck", [](API_ARGS_MAP) {
         CHECK_SECRET();
         CHECK_ARGS("stream_id");
-        auto rtp_process = RtpSelector::Instance().getProcess(allArgs["stream_id"], false);
+        auto src = MediaSource::find(DEFAULT_VHOST, kRtpAppName, allArgs["stream_id"]);
-        if (rtp_process) {
+        auto process = src ? src->getRtpProcess() : nullptr;
-            rtp_process->setStopCheckRtp(false);
+        if (process) {
+            process->setStopCheckRtp(false);
         } else {
             val["code"] = API::NotFound;
         }
@@ -1588,12 +1597,14 @@ void installWebApi() {
         auto record_path = Recorder::getRecordPath(Recorder::type_mp4, tuple, allArgs["customized_path"]);
         auto period = allArgs["period"];
         record_path = record_path + period + "/";
+
+        bool recording = false;
         auto name = allArgs["name"];
         if (!name.empty()) {
+            // 删除指定文件
             record_path += name;
-        }
+        } else {
-        bool recording = false;
+            // 删除文件夹，先判断该流是否正在录制中
-        {
             auto src = MediaSource::find(allArgs["vhost"], allArgs["app"], allArgs["stream"]);
             if (src && src->isRecording(Recorder::type_mp4)) {
                 recording = true;
@@ -1762,7 +1773,7 @@ void installWebApi() {
             , _session_id(std::move(session_id)) {}
         ~WebRtcArgsImp() override = default;
 
-        variant operator[](const string &key) const override {
+        toolkit::variant operator[](const string &key) const override {
             if (key == "url") {
                 return getUrl();
             }

src/Codec/H264Encoder.cpp

@@ -321,6 +321,8 @@ int H264Encoder::inputData(char *yuv[3], int linesize[3], int64_t cts, H264Frame
         _aFrames[i].iType = pNal.i_type;
         _aFrames[i].iLength = pNal.i_payload;
         _aFrames[i].pucData = pNal.p_payload;
+        _aFrames[i].dts = _pPicOut->i_dts;
+        _aFrames[i].pts = _pPicOut->i_pts;
     }
     *out_frame = _aFrames;
     return iNal;

src/Codec/H264Encoder.h

@@ -27,6 +27,9 @@ public:
         int iType;
         int iLength;
         uint8_t *pucData;
+
+        int64_t dts;
+        int64_t pts;
     } H264Frame;
 
     H264Encoder();

src/Common/Device.cpp

@@ -39,7 +39,7 @@ bool DevChannel::inputYUV(char *yuv[3], int linesize[3], uint64_t cts) {
         int frames = _pH264Enc->inputData(yuv, linesize, cts, &out_frames);
         bool ret = false;
         for (int i = 0; i < frames; i++) {
-            ret = inputH264((char *) out_frames[i].pucData, out_frames[i].iLength, cts) ? true : ret;
+            ret = inputH264((char *) out_frames[i].pucData, out_frames[i].iLength, out_frames[i].dts, out_frames[i].pts) ? true : ret;
         }
         return ret;
     }

src/Common/MediaSource.cpp

@@ -271,9 +271,14 @@ toolkit::EventPoller::Ptr MediaSource::getOwnerPoller() {
     throw std::runtime_error(toolkit::demangle(typeid(*this).name()) + "::getOwnerPoller failed: " + getUrl());
 }
 
-std::shared_ptr<MultiMediaSourceMuxer> MediaSource::getMuxer() {
+std::shared_ptr<MultiMediaSourceMuxer> MediaSource::getMuxer() const {
     auto listener = _listener.lock();
-    return listener ? listener->getMuxer(*this) : nullptr;
+    return listener ? listener->getMuxer(const_cast<MediaSource&>(*this)) : nullptr;
+}
+
+std::shared_ptr<RtpProcess> MediaSource::getRtpProcess() const {
+    auto listener = _listener.lock();
+    return listener ? listener->getRtpProcess(const_cast<MediaSource&>(*this)) : nullptr;
 }
 
 void MediaSource::onReaderChanged(int size) {
@@ -652,6 +657,10 @@ MediaSource::Ptr MediaSource::createFromMP4(const string &schema, const string &
 /////////////////////////////////////MediaSourceEvent//////////////////////////////////////
 
 void MediaSourceEvent::onReaderChanged(MediaSource &sender, int size){
+    GET_CONFIG(bool, enable, General::kBroadcastPlayerCountChanged);
+    if (enable) {
+        NOTICE_EMIT(BroadcastPlayerCountChangedArgs, Broadcast::kBroadcastPlayerCountChanged, sender.getMediaTuple(), sender.totalReaderCount());
+    }
     if (size || sender.totalReaderCount()) {
         //还有人观看该视频，不触发关闭事件
         _async_close_timer = nullptr;
@@ -799,11 +808,16 @@ toolkit::EventPoller::Ptr MediaSourceEventInterceptor::getOwnerPoller(MediaSourc
     throw std::runtime_error(toolkit::demangle(typeid(*this).name()) + "::getOwnerPoller failed");
 }
 
-std::shared_ptr<MultiMediaSourceMuxer> MediaSourceEventInterceptor::getMuxer(MediaSource &sender) {
+std::shared_ptr<MultiMediaSourceMuxer> MediaSourceEventInterceptor::getMuxer(MediaSource &sender) const {
     auto listener = _listener.lock();
     return listener ? listener->getMuxer(sender) : nullptr;
 }
 
+std::shared_ptr<RtpProcess> MediaSourceEventInterceptor::getRtpProcess(MediaSource &sender) const {
+    auto listener = _listener.lock();
+    return listener ? listener->getRtpProcess(sender) : nullptr;
+}
+
 bool MediaSourceEventInterceptor::setupRecord(MediaSource &sender, Recorder::type type, bool start, const string &custom_path, size_t max_second) {
     auto listener = _listener.lock();
     if (!listener) {

src/Common/MediaSource.h

@@ -41,6 +41,7 @@ enum class MediaOriginType : uint8_t {
 std::string getOriginTypeString(MediaOriginType type);
 
 class MediaSource;
+class RtpProcess;
 class MultiMediaSourceMuxer;
 class MediaSourceEvent {
 public:
@@ -88,7 +89,9 @@ public:
     // 获取所有track相关信息
     virtual std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const { return std::vector<Track::Ptr>(); };
     // 获取MultiMediaSourceMuxer对象
-    virtual std::shared_ptr<MultiMediaSourceMuxer> getMuxer(MediaSource &sender) { return nullptr; }
+    virtual std::shared_ptr<MultiMediaSourceMuxer> getMuxer(MediaSource &sender) const { return nullptr; }
+    // 获取RtpProcess对象
+    virtual std::shared_ptr<RtpProcess> getRtpProcess(MediaSource &sender) const { return nullptr; }
 
     class SendRtpArgs {
     public:
@@ -278,7 +281,8 @@ public:
     bool stopSendRtp(MediaSource &sender, const std::string &ssrc) override;
     float getLossRate(MediaSource &sender, TrackType type) override;
     toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override;
-    std::shared_ptr<MultiMediaSourceMuxer> getMuxer(MediaSource &sender) override;
+    std::shared_ptr<MultiMediaSourceMuxer> getMuxer(MediaSource &sender) const override;
+    std::shared_ptr<RtpProcess> getRtpProcess(MediaSource &sender) const override;
 
 private:
     std::weak_ptr<MediaSourceEvent> _listener;
@@ -395,7 +399,9 @@ public:
     // 获取所在线程
     toolkit::EventPoller::Ptr getOwnerPoller();
     // 获取MultiMediaSourceMuxer对象
-    std::shared_ptr<MultiMediaSourceMuxer> getMuxer();
+    std::shared_ptr<MultiMediaSourceMuxer> getMuxer() const;
+    // 获取RtpProcess对象
+    std::shared_ptr<RtpProcess> getRtpProcess() const;
 
     ////////////////static方法，查找或生成MediaSource////////////////
 

src/Common/MultiMediaSourceMuxer.cpp

@@ -470,8 +470,8 @@ EventPoller::Ptr MultiMediaSourceMuxer::getOwnerPoller(MediaSource &sender) {
     }
 }
 
-std::shared_ptr<MultiMediaSourceMuxer> MultiMediaSourceMuxer::getMuxer(MediaSource &sender) {
+std::shared_ptr<MultiMediaSourceMuxer> MultiMediaSourceMuxer::getMuxer(MediaSource &sender) const {
-    return shared_from_this();
+    return const_cast<MultiMediaSourceMuxer*>(this)->shared_from_this();
 }
 
 bool MultiMediaSourceMuxer::onTrackReady(const Track::Ptr &track) {

src/Common/MultiMediaSourceMuxer.h

@@ -132,7 +132,7 @@ public:
     /**
      * 获取本对象
      */
-    std::shared_ptr<MultiMediaSourceMuxer> getMuxer(MediaSource &sender) override;
+    std::shared_ptr<MultiMediaSourceMuxer> getMuxer(MediaSource &sender) const override;
 
     const ProtocolOption &getOption() const;
     const MediaTuple &getMediaTuple() const;

src/Common/config.cpp

@@ -64,6 +64,7 @@ const string kBroadcastRtcSctpFailed = "kBroadcastRtcSctpFailed";
 const string kBroadcastRtcSctpClosed = "kBroadcastRtcSctpClosed";
 const string kBroadcastRtcSctpSend = "kBroadcastRtcSctpSend";
 const string kBroadcastRtcSctpReceived = "kBroadcastRtcSctpReceived";
+const string kBroadcastPlayerCountChanged = "kBroadcastPlayerCountChanged";
 
 } // namespace Broadcast
 
@@ -82,6 +83,7 @@ const string kEnableFFmpegLog = GENERAL_FIELD "enable_ffmpeg_log";
 const string kWaitTrackReadyMS = GENERAL_FIELD "wait_track_ready_ms";
 const string kWaitAddTrackMS = GENERAL_FIELD "wait_add_track_ms";
 const string kUnreadyFrameCache = GENERAL_FIELD "unready_frame_cache";
+const string kBroadcastPlayerCountChanged = GENERAL_FIELD "broadcast_player_count_changed";
 
 static onceToken token([]() {
     mINI::Instance()[kFlowThreshold] = 1024;
@@ -96,6 +98,7 @@ static onceToken token([]() {
     mINI::Instance()[kWaitTrackReadyMS] = 10000;
     mINI::Instance()[kWaitAddTrackMS] = 3000;
     mINI::Instance()[kUnreadyFrameCache] = 100;
+    mINI::Instance()[kBroadcastPlayerCountChanged] = 0;
 });
 
 } // namespace General
@@ -361,6 +364,7 @@ static onceToken token([]() {
 namespace Client {
 const string kNetAdapter = "net_adapter";
 const string kRtpType = "rtp_type";
+const string kRtspBeatType = "rtsp_beat_type";
 const string kRtspUser = "rtsp_user";
 const string kRtspPwd = "rtsp_pwd";
 const string kRtspPwdIsMD5 = "rtsp_pwd_md5";

src/Common/config.h

@@ -124,6 +124,10 @@ extern const std::string kBroadcastRtcSctpSend;
 extern const std::string kBroadcastRtcSctpReceived;
 #define BroadcastRtcSctpReceivedArgs WebRtcTransport& sender, uint16_t &streamId, uint32_t &ppid, const uint8_t *&msg, size_t &len
 
+// 观看人数变化广播
+extern const std::string kBroadcastPlayerCountChanged;
+#define BroadcastPlayerCountChangedArgs const MediaTuple& args, const int& count
+
 #define ReloadConfigTag ((void *)(0xFF))
 #define RELOAD_KEY(arg, key)                                                                                           \
     do {                                                                                                               \
@@ -196,6 +200,8 @@ extern const std::string kWaitTrackReadyMS;
 extern const std::string kWaitAddTrackMS;
 // 如果track未就绪，我们先缓存帧数据，但是有最大个数限制(100帧时大约4秒)，防止内存溢出
 extern const std::string kUnreadyFrameCache;
+// 是否启用观看人数变化事件广播，置1则启用，置0则关闭
+extern const std::string kBroadcastPlayerCountChanged;
 } // namespace General
 
 namespace Protocol {
@@ -417,6 +423,9 @@ extern const std::string kNetAdapter;
 // 设置rtp传输类型，可选项有0(tcp，默认)、1(udp)、2(组播)
 // 设置方法:player[PlayerBase::kRtpType] = 0/1/2;
 extern const std::string kRtpType;
+// rtsp播放器发送信令心跳还是rtcp心跳，可选项有0(同时发)、1(rtcp心跳)、2(信令心跳)
+// 设置方法:player[PlayerBase::kRtspBeatType] = 0/1/2;
+extern const std::string kRtspBeatType;
 // rtsp认证用户名
 extern const std::string kRtspUser;
 // rtsp认证用用户密码，可以是明文也可以是md5,md5密码生成方式 md5(username:realm:password)

src/Record/Recorder.h

@@ -32,7 +32,7 @@ class RecordInfo: public MediaTuple {
 public:
     time_t start_time;  // GMT 标准时间，单位秒
     float time_len;     // 录像长度，单位秒
-    off_t file_size;    // 文件大小，单位 BYTE
+    uint64_t file_size;    // 文件大小，单位 BYTE
     std::string file_path;   // 文件路径
     std::string file_name;   // 文件名称
     std::string folder;      // 文件夹路径

src/Rtp/RtpProcess.cpp

@@ -11,26 +11,29 @@
 #if defined(ENABLE_RTPPROXY)
 #include "GB28181Process.h"
 #include "RtpProcess.h"
-#include "RtpSelector.h"
-#include "Http/HttpTSPlayer.h"
 #include "Util/File.h"
 #include "Common/config.h"
 
 using namespace std;
 using namespace toolkit;
 
-static constexpr char kRtpAppName[] = "rtp";
 //在创建_muxer对象前(也就是推流鉴权成功前)，需要先缓存frame，这样可以防止丢包，提高体验
 //但是同时需要控制缓冲长度，防止内存溢出。200帧数据，大概有10秒数据，应该足矣等待鉴权hook返回
 static constexpr size_t kMaxCachedFrame = 200;
 
 namespace mediakit {
 
-RtpProcess::RtpProcess(const string &stream_id) {
+RtpProcess::Ptr RtpProcess::createProcess(std::string stream_id) {
+     RtpProcess::Ptr ret(new RtpProcess(std::move(stream_id)));
+     ret->createTimer();
+     return ret;
+}
+
+RtpProcess::RtpProcess(string stream_id) {
     _media_info.schema = kRtpAppName;
     _media_info.vhost = DEFAULT_VHOST;
     _media_info.app = kRtpAppName;
-    _media_info.stream = stream_id;
+    _media_info.stream = std::move(stream_id);
 
     GET_CONFIG(string, dump_dir, RtpProxy::kDumpDir);
     {
@@ -75,6 +78,25 @@ RtpProcess::~RtpProcess() {
     }
 }
 
+void RtpProcess::onManager() {
+    if (!alive()) {
+        onDetach(SockException(Err_timeout, "RtpProcess timeout"));
+    }
+}
+
+void RtpProcess::createTimer() {
+    //创建超时管理定时器
+    weak_ptr<RtpProcess> weakSelf = shared_from_this();
+    _timer = std::make_shared<Timer>(3.0f, [weakSelf] {
+        auto strongSelf = weakSelf.lock();
+        if (!strongSelf) {
+            return false;
+        }
+        strongSelf->onManager();
+        return true;
+    }, EventPollerPool::Instance().getPoller());
+}
+
 bool RtpProcess::inputRtp(bool is_udp, const Socket::Ptr &sock, const char *data, size_t len, const struct sockaddr *addr, uint64_t *dts_out) {
     if (!isRtp(data, len)) {
         WarnP(this) << "Not rtp packet";
@@ -203,13 +225,14 @@ void RtpProcess::setOnlyTrack(OnlyTrack only_track) {
     _only_track = only_track;
 }
 
-void RtpProcess::onDetach() {
+void RtpProcess::onDetach(const SockException &ex) {
     if (_on_detach) {
-        _on_detach();
+        WarnL << ex << ", stream_id: " << getIdentifier();
+        _on_detach(ex);
     }
 }
 
-void RtpProcess::setOnDetach(function<void()> cb) {
+void RtpProcess::setOnDetach(onDetachCB cb) {
     _on_detach = std::move(cb);
 }
 
@@ -256,9 +279,6 @@ void RtpProcess::emitOnPublish() {
             }
             if (err.empty()) {
                 strong_self->_muxer = std::make_shared<MultiMediaSourceMuxer>(strong_self->_media_info, 0.0f, option);
-                if (!option.stream_replace.empty()) {
-                    RtpSelector::Instance().addStreamReplace(strong_self->_media_info.stream, option.stream_replace);
-                }
                 switch (strong_self->_only_track) {
                     case kOnlyAudio: strong_self->_muxer->setOnlyAudio(); break;
                     case kOnlyVideo: strong_self->_muxer->enableAudio(false); break;
@@ -294,6 +314,15 @@ std::shared_ptr<SockInfo> RtpProcess::getOriginSock(MediaSource &sender) const {
     return const_cast<RtpProcess *>(this)->shared_from_this();
 }
 
+RtpProcess::Ptr RtpProcess::getRtpProcess(mediakit::MediaSource &sender) const {
+    return const_cast<RtpProcess *>(this)->shared_from_this();
+}
+
+bool RtpProcess::close(mediakit::MediaSource &sender) {
+    onDetach(SockException(Err_shutdown, "close media"));
+    return true;
+}
+
 toolkit::EventPoller::Ptr RtpProcess::getOwnerPoller(MediaSource &sender) {
     if (_sock) {
         return _sock->getPoller();

src/Rtp/RtpProcess.h

@@ -18,11 +18,14 @@
 
 namespace mediakit {
 
-class RtpProcess final : public RtcpContextForRecv, public toolkit::SockInfo, public MediaSinkInterface, public MediaSourceEventInterceptor, public std::enable_shared_from_this<RtpProcess>{
+static constexpr char kRtpAppName[] = "rtp";
+
+class RtpProcess final : public RtcpContextForRecv, public toolkit::SockInfo, public MediaSinkInterface, public MediaSourceEvent, public std::enable_shared_from_this<RtpProcess>{
 public:
     using Ptr = std::shared_ptr<RtpProcess>;
-    friend class RtpProcessHelper;
+    using onDetachCB = std::function<void(const toolkit::SockException &ex)>;
-    RtpProcess(const std::string &stream_id);
+
+    static Ptr createProcess(std::string stream_id);
     ~RtpProcess();
     enum OnlyTrack { kAll = 0, kOnlyAudio = 1, kOnlyVideo = 2 };
 
@@ -38,20 +41,16 @@ public:
      */
     bool inputRtp(bool is_udp, const toolkit::Socket::Ptr &sock, const char *data, size_t len, const struct sockaddr *addr , uint64_t *dts_out = nullptr);
 
-    /**
-     * 是否超时，用于超时移除对象
-     */
-    bool alive();
 
     /**
      * 超时时被RtpSelector移除时触发
      */
-    void onDetach();
+    void onDetach(const toolkit::SockException &ex);
 
     /**
      * 设置onDetach事件回调
      */
-    void setOnDetach(std::function<void()> cb);
+    void setOnDetach(onDetachCB cb);
 
     /**
      * 设置onDetach事件回调,false检查RTP超时，true停止
@@ -88,10 +87,17 @@ protected:
     std::shared_ptr<SockInfo> getOriginSock(MediaSource &sender) const override;
     toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override;
     float getLossRate(MediaSource &sender, TrackType type) override;
+    Ptr getRtpProcess(mediakit::MediaSource &sender) const override;
+    bool close(mediakit::MediaSource &sender) override;
 
 private:
+    RtpProcess(std::string stream_id);
+
     void emitOnPublish();
     void doCachedFunc();
+    bool alive();
+    void onManager();
+    void createTimer();
 
 private:
     OnlyTrack _only_track = kAll;
@@ -102,12 +108,13 @@ private:
     toolkit::Socket::Ptr _sock;
     MediaInfo _media_info;
     toolkit::Ticker _last_frame_time;
-    std::function<void()> _on_detach;
+    onDetachCB _on_detach;
     std::shared_ptr<FILE> _save_file_rtp;
     std::shared_ptr<FILE> _save_file_video;
     ProcessInterface::Ptr _process;
     MultiMediaSourceMuxer::Ptr _muxer;
     std::atomic_bool _stop_rtp_check{false};
+    toolkit::Timer::Ptr _timer;
     toolkit::Ticker _last_check_alive;
     std::recursive_mutex _func_mtx;
     std::deque<std::function<void()> > _cached_func;

src/Rtp/RtpSelector.cpp

@@ -1,168 +0,0 @@
-/*
- * 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.
- */
-
-#if defined(ENABLE_RTPPROXY)
-#include <stddef.h>
-#include "RtpSelector.h"
-#include "RtpSplitter.h"
-
-using namespace std;
-using namespace toolkit;
-
-namespace mediakit{
-
-INSTANCE_IMP(RtpSelector);
-
-void RtpSelector::clear(){
-    lock_guard<decltype(_mtx_map)> lck(_mtx_map);
-    _map_rtp_process.clear();
-    _map_stream_replace.clear();
-}
-
-bool RtpSelector::getSSRC(const char *data, size_t data_len, uint32_t &ssrc){
-    if (data_len < 12) {
-        return false;
-    }
-    uint32_t *ssrc_ptr = (uint32_t *) (data + 8);
-    ssrc = ntohl(*ssrc_ptr);
-    return true;
-}
-
-RtpProcess::Ptr RtpSelector::getProcess(const string &stream_id,bool makeNew) {
-    lock_guard<decltype(_mtx_map)> lck(_mtx_map);
-    string stream_id_origin = stream_id;
-    auto it_replace = _map_stream_replace.find(stream_id);
-    if (it_replace != _map_stream_replace.end()) {
-        stream_id_origin = it_replace->second;
-    }
-
-    auto it = _map_rtp_process.find(stream_id_origin);
-    if (it == _map_rtp_process.end() && !makeNew) {
-        return nullptr;
-    }
-    if (it != _map_rtp_process.end() && makeNew) {
-        //已经被其他线程持有了，不得再被持有，否则会存在线程安全的问题
-        throw ProcessExisted(StrPrinter << "RtpProcess(" << stream_id_origin << ") already existed");
-    }
-    RtpProcessHelper::Ptr &ref = _map_rtp_process[stream_id_origin];
-    if (!ref) {
-        ref = std::make_shared<RtpProcessHelper>(stream_id_origin, shared_from_this());
-        ref->attachEvent();
-        createTimer();
-    }
-    return ref->getProcess();
-}
-
-void RtpSelector::createTimer() {
-    if (!_timer) {
-        //创建超时管理定时器
-        weak_ptr<RtpSelector> weakSelf = shared_from_this();
-        _timer = std::make_shared<Timer>(3.0f, [weakSelf] {
-            auto strongSelf = weakSelf.lock();
-            if (!strongSelf) {
-                return false;
-            }
-            strongSelf->onManager();
-            return true;
-        }, EventPollerPool::Instance().getPoller());
-    }
-}
-
-void RtpSelector::delProcess(const string &stream_id,const RtpProcess *ptr) {
-    RtpProcess::Ptr process;
-    {
-        lock_guard<decltype(_mtx_map)> lck(_mtx_map);
-        auto it = _map_rtp_process.find(stream_id);
-        if (it == _map_rtp_process.end()) {
-            return;
-        }
-        if (it->second->getProcess().get() != ptr) {
-            return;
-        }
-        process = it->second->getProcess();
-        _map_rtp_process.erase(it);
-        delStreamReplace(stream_id);
-    }
-    process->onDetach();
-}
-
-void RtpSelector::addStreamReplace(const string &stream_id, const std::string &stream_replace) {
-    lock_guard<decltype(_mtx_map)> lck(_mtx_map);
-    _map_stream_replace[stream_replace] = stream_id;
-}
-
-void RtpSelector::delStreamReplace(const string &stream_id) {
-    for (auto it = _map_stream_replace.begin(); it != _map_stream_replace.end(); ++it) {
-        if (it->second == stream_id) {
-            _map_stream_replace.erase(it);
-            break;
-        }
-    }
-}
-
-void RtpSelector::onManager() {
-    List<RtpProcess::Ptr> clear_list;
-    {
-        lock_guard<decltype(_mtx_map)> lck(_mtx_map);
-        for (auto it = _map_rtp_process.begin(); it != _map_rtp_process.end();) {
-            if (it->second->getProcess()->alive()) {
-                ++it;
-                continue;
-            }
-            WarnL << "RtpProcess timeout:" << it->first;
-            clear_list.emplace_back(it->second->getProcess());
-            delStreamReplace(it->first);
-            it = _map_rtp_process.erase(it);
-        }
-    }
-
-    clear_list.for_each([](const RtpProcess::Ptr &process) {
-        process->onDetach();
-    });
-}
-
-RtpProcessHelper::RtpProcessHelper(const string &stream_id, const weak_ptr<RtpSelector> &parent) {
-    _stream_id = stream_id;
-    _parent = parent;
-    _process = std::make_shared<RtpProcess>(stream_id);
-}
-
-RtpProcessHelper::~RtpProcessHelper() {
-    auto process = std::move(_process);
-    try {
-        // flush时，确保线程安全
-        process->getOwnerPoller(MediaSource::NullMediaSource())->async([process]() { process->flush(); });
-    } catch (...) {
-        // 忽略getOwnerPoller可能抛出的异常
-    }
-}
-
-void RtpProcessHelper::attachEvent() {
-    //主要目的是close回调触发时能把对象从RtpSelector中删除
-    _process->setDelegate(shared_from_this());
-}
-
-bool RtpProcessHelper::close(MediaSource &sender) {
-    //此回调在其他线程触发
-    auto parent = _parent.lock();
-    if (!parent) {
-        return false;
-    }
-    parent->delProcess(_stream_id, _process.get());
-    WarnL << "close media: " << sender.getUrl();
-    return true;
-}
-
-RtpProcess::Ptr &RtpProcessHelper::getProcess() {
-    return _process;
-}
-
-}//namespace mediakit
-#endif//defined(ENABLE_RTPPROXY)

src/Rtp/RtpSelector.h

@@ -1,89 +0,0 @@
-/*
- * 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.
- */
-
-#ifndef ZLMEDIAKIT_RTPSELECTOR_H
-#define ZLMEDIAKIT_RTPSELECTOR_H
-
-#if defined(ENABLE_RTPPROXY)
-#include <stdint.h>
-#include <mutex>
-#include <unordered_map>
-#include "RtpProcess.h"
-#include "Common/MediaSource.h"
-
-namespace mediakit{
-
-class RtpSelector;
-class RtpProcessHelper : public MediaSourceEvent , public std::enable_shared_from_this<RtpProcessHelper> {
-public:
-    using Ptr = std::shared_ptr<RtpProcessHelper>;
-    RtpProcessHelper(const std::string &stream_id, const std::weak_ptr<RtpSelector > &parent);
-    ~RtpProcessHelper();
-    void attachEvent();
-    RtpProcess::Ptr & getProcess();
-
-protected:
-    // 通知其停止推流
-    bool close(MediaSource &sender) override;
-
-private:
-    std::string _stream_id;
-    RtpProcess::Ptr _process;
-    std::weak_ptr<RtpSelector> _parent;
-};
-
-class RtpSelector : public std::enable_shared_from_this<RtpSelector>{
-public:
-    class ProcessExisted : public std::runtime_error {
-    public:
-        template<typename ...T>
-        ProcessExisted(T && ...args) : std::runtime_error(std::forward<T>(args)...) {}
-    };
-
-    static bool getSSRC(const char *data,size_t data_len, uint32_t &ssrc);
-    static RtpSelector &Instance();
-
-    /**
-     * 清空所有对象
-     */
-    void clear();
-
-    /**
-     * 获取一个rtp处理器
-     * @param stream_id 流id
-     * @param makeNew 不存在时是否新建, 该参数为true时，必须确保之前未创建同名对象
-     * @return rtp处理器
-     */
-    RtpProcess::Ptr getProcess(const std::string &stream_id, bool makeNew);
-
-    /**
-     * 删除rtp处理器
-     * @param stream_id 流id
-     * @param ptr rtp处理器指针
-     */
-    void delProcess(const std::string &stream_id, const RtpProcess *ptr);
-
-    void addStreamReplace(const std::string &stream_id, const std::string &stream_replace);
-
-private:
-    void onManager();
-    void createTimer();
-    void delStreamReplace(const std::string &stream_id);
-
-private:
-    toolkit::Timer::Ptr _timer;
-    std::recursive_mutex _mtx_map;
-    std::unordered_map<std::string,RtpProcessHelper::Ptr> _map_rtp_process;
-    std::unordered_map<std::string,std::string> _map_stream_replace;
-};
-
-}//namespace mediakit
-#endif//defined(ENABLE_RTPPROXY)
-#endif //ZLMEDIAKIT_RTPSELECTOR_H

src/Rtp/RtpServer.cpp

@@ -11,7 +11,7 @@
 #if defined(ENABLE_RTPPROXY)
 #include "Util/uv_errno.h"
 #include "RtpServer.h"
-#include "RtpSelector.h"
+#include "RtpProcess.h"
 #include "Rtcp/RtcpContext.h"
 #include "Common/config.h"
 
@@ -35,38 +35,34 @@ public:
         _stream_id = std::move(stream_id);
     }
 
-    ~RtcpHelper() {
-        if (_process) {
-            // 删除rtp处理器
-            RtpSelector::Instance().delProcess(_stream_id, _process.get());
-        }
-    }
-
     void setRtpServerInfo(uint16_t local_port, RtpServer::TcpMode mode, bool re_use_port, uint32_t ssrc, int only_track) {
-        _local_port = local_port;
-        _tcp_mode = mode;
-        _re_use_port = re_use_port;
         _ssrc = ssrc;
-        _only_track = only_track;
+        _process = RtpProcess::createProcess(_stream_id);
+        _process->setOnlyTrack((RtpProcess::OnlyTrack)only_track);
+
+        _timeout_cb = [=]() mutable {
+            NOTICE_EMIT(BroadcastRtpServerTimeoutArgs, Broadcast::kBroadcastRtpServerTimeout, local_port, _stream_id, (int)mode, re_use_port, ssrc);
+        };
+
+        weak_ptr<RtcpHelper> weak_self = shared_from_this();
+        _process->setOnDetach([weak_self](const SockException &ex) {
+            if (auto strong_self = weak_self.lock()) {
+                if (strong_self->_on_detach) {
+                    strong_self->_on_detach(ex);
+                }
+                if (ex.getErrCode() == Err_timeout) {
+                    strong_self->_timeout_cb();
+                }
+            }
+        });
     }
 
-    void setOnDetach(function<void()> cb) {
+    void setOnDetach(RtpProcess::onDetachCB cb) { _on_detach = std::move(cb); }
-        if (_process) {
+
-            _process->setOnDetach(std::move(cb));
+    RtpProcess::Ptr getProcess() const { return _process; }
-        } else {
-            _on_detach = std::move(cb);
-        }
-    }
 
     void onRecvRtp(const Socket::Ptr &sock, const Buffer::Ptr &buf, struct sockaddr *addr) {
-        if (!_process) {
-            _process = RtpSelector::Instance().getProcess(_stream_id, true);
-            _process->setOnlyTrack((RtpProcess::OnlyTrack)_only_track);
-            _process->setOnDetach(std::move(_on_detach));
-            cancelDelayTask();
-        }
         _process->inputRtp(true, sock, buf->data(), buf->size(), addr);
-
         // 统计rtp接受情况，用于发送rr包
         auto header = (RtpHeader *)buf->data();
         sendRtcp(ntohl(header->ssrc), addr);
@@ -92,37 +88,12 @@ public:
             // 收到sr rtcp后驱动返回rr rtcp
             strong_self->sendRtcp(strong_self->_ssrc, (struct sockaddr *)(strong_self->_rtcp_addr.get()));
         });
-
-        GET_CONFIG(uint64_t, timeoutSec, RtpProxy::kTimeoutSec);
-        _delay_task = _rtcp_sock->getPoller()->doDelayTask(timeoutSec * 1000, [weak_self]() {
-            if (auto strong_self = weak_self.lock()) {
-                auto process = RtpSelector::Instance().getProcess(strong_self->_stream_id, false);
-                if (!process && strong_self->_on_detach) {
-                    strong_self->_on_detach();
-                }
-                if(process && strong_self->_on_detach){// tcp 链接防止断开不删除rtpServer
-                    process->setOnDetach(std::move(strong_self->_on_detach));
-                }
-                if (!process) { // process 未创建，触发rtp server 超时事件
-                    NOTICE_EMIT(BroadcastRtpServerTimeoutArgs, Broadcast::kBroadcastRtpServerTimeout, strong_self->_local_port, strong_self->_stream_id,
-                                (int)strong_self->_tcp_mode, strong_self->_re_use_port, strong_self->_ssrc);
-                }
-            }
-            return 0;
-        });
-    }
-
-    void cancelDelayTask() {
-        if (_delay_task) {
-            _delay_task->cancel();
-            _delay_task = nullptr;
-        }
     }
 
 private:
     void sendRtcp(uint32_t rtp_ssrc, struct sockaddr *addr) {
         // 每5秒发送一次rtcp
-        if (_ticker.elapsedTime() < 5000 || !_process) {
+        if (_ticker.elapsedTime() < 5000) {
             return;
         }
         _ticker.resetTime();
@@ -141,19 +112,14 @@ private:
     }
 
 private:
-    bool _re_use_port = false;
-    int _only_track = 0;
-    uint16_t _local_port = 0;
     uint32_t _ssrc = 0;
-    RtpServer::TcpMode _tcp_mode = RtpServer::NONE;
+    std::function<void()> _timeout_cb;
-
     Ticker _ticker;
     Socket::Ptr _rtcp_sock;
     RtpProcess::Ptr _process;
     std::string _stream_id;
-    function<void()> _on_detach;
+    RtpProcess::onDetachCB _on_detach;
     std::shared_ptr<struct sockaddr_storage> _rtcp_addr;
-    EventPoller::DelayTask::Ptr _delay_task;
 };
 
 void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_mode, const char *local_ip, bool re_use_port, uint32_t ssrc, int only_track, bool multiplex) {
@@ -177,22 +143,6 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
     GET_CONFIG(int, udpRecvSocketBuffer, RtpProxy::kUdpRecvSocketBuffer);
     SockUtil::setRecvBuf(rtp_socket->rawFD(), udpRecvSocketBuffer);
 
-    TcpServer::Ptr tcp_server;
-    _tcp_mode = tcp_mode;
-    if (tcp_mode == PASSIVE || tcp_mode == ACTIVE) {
-        //创建tcp服务器
-        tcp_server = std::make_shared<TcpServer>(rtp_socket->getPoller());
-        (*tcp_server)[RtpSession::kStreamID] = stream_id;
-        (*tcp_server)[RtpSession::kSSRC] = ssrc;
-        (*tcp_server)[RtpSession::kOnlyTrack] = only_track;
-        if (tcp_mode == PASSIVE) {
-            tcp_server->start<RtpSession>(local_port, local_ip);
-        } else if (stream_id.empty()) {
-            // tcp主动模式时只能一个端口一个流，必须指定流id; 创建TcpServer对象也仅用于传参
-            throw std::runtime_error(StrPrinter << "tcp主动模式时必需指定流id");
-        }
-    }
-
     //创建udp服务器
     UdpServer::Ptr udp_server;
     RtcpHelper::Ptr helper;
@@ -222,13 +172,32 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
         });
     } else {
         //单端口多线程接收多个流，根据ssrc区分流
-        udp_server = std::make_shared<UdpServer>(rtp_socket->getPoller());
+        udp_server = std::make_shared<UdpServer>();
         (*udp_server)[RtpSession::kOnlyTrack] = only_track;
         (*udp_server)[RtpSession::kUdpRecvBuffer] = udpRecvSocketBuffer;
         udp_server->start<RtpSession>(local_port, local_ip);
         rtp_socket = nullptr;
     }
 
+    TcpServer::Ptr tcp_server;
+    if (tcp_mode == PASSIVE || tcp_mode == ACTIVE) {
+        //创建tcp服务器
+        tcp_server = std::make_shared<TcpServer>();
+        (*tcp_server)[RtpSession::kStreamID] = stream_id;
+        (*tcp_server)[RtpSession::kSSRC] = ssrc;
+        (*tcp_server)[RtpSession::kOnlyTrack] = only_track;
+        if (tcp_mode == PASSIVE) {
+            weak_ptr<RtpServer> weak_self = shared_from_this();
+            auto processor = helper ? helper->getProcess() : nullptr;
+            tcp_server->start<RtpSession>(local_port, local_ip, 1024, [weak_self, processor](std::shared_ptr<RtpSession> &session) {
+                session->setRtpProcess(processor);
+            });
+        } else if (stream_id.empty()) {
+            // tcp主动模式时只能一个端口一个流，必须指定流id; 创建TcpServer对象也仅用于传参
+            throw std::runtime_error(StrPrinter << "tcp主动模式时必需指定流id");
+        }
+    }
+
     _on_cleanup = [rtp_socket, stream_id]() {
         if (rtp_socket) {
             //去除循环引用
@@ -240,9 +209,10 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
     _udp_server = udp_server;
     _rtp_socket = rtp_socket;
     _rtcp_helper = helper;
+    _tcp_mode = tcp_mode;
 }
 
-void RtpServer::setOnDetach(function<void()> cb) {
+void RtpServer::setOnDetach(RtpProcess::onDetachCB cb) {
     if (_rtcp_helper) {
         _rtcp_helper->setOnDetach(std::move(cb));
     }
@@ -277,6 +247,7 @@ void RtpServer::connectToServer(const std::string &url, uint16_t port, const fun
 
 void RtpServer::onConnect() {
     auto rtp_session = std::make_shared<RtpSession>(_rtp_socket);
+    rtp_session->setRtpProcess(_rtcp_helper->getProcess());
     rtp_session->attachServer(*_tcp_server);
     _rtp_socket->setOnRead([rtp_session](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) {
         rtp_session->onRecv(buf);

src/Rtp/RtpServer.h

@@ -62,7 +62,7 @@ public:
     /**
      * 设置RtpProcess onDetach事件回调
      */
-    void setOnDetach(std::function<void()> cb);
+    void setOnDetach(RtpProcess::onDetachCB cb);
 
     /**
      * 更新ssrc

src/Rtp/RtpSession.cpp

@@ -10,7 +10,7 @@
 
 #if defined(ENABLE_RTPPROXY)
 #include "RtpSession.h"
-#include "RtpSelector.h"
+#include "RtpProcess.h"
 #include "Network/TcpServer.h"
 #include "Rtsp/Rtsp.h"
 #include "Rtsp/RtpReceiver.h"
@@ -60,28 +60,24 @@ void RtpSession::onRecv(const Buffer::Ptr &data) {
 }
 
 void RtpSession::onError(const SockException &err) {
-    WarnP(this) << _stream_id << " " << err;
+    if (_emit_detach) {
-    if (_process) {
+        _process->onDetach(err);
-        RtpSelector::Instance().delProcess(_stream_id, _process.get());
-        _process = nullptr;
     }
+    WarnP(this) << _stream_id << " " << err;
 }
 
 void RtpSession::onManager() {
-    if (_process && !_process->alive()) {
-        shutdown(SockException(Err_timeout, "receive rtp timeout"));
-    }
-
     if (!_process && _ticker.createdTime() > 10 * 1000) {
         shutdown(SockException(Err_timeout, "illegal connection"));
     }
 }
 
+void RtpSession::setRtpProcess(RtpProcess::Ptr process) {
+    _emit_detach = (bool)process;
+    _process = std::move(process);
+}
+
 void RtpSession::onRtpPacket(const char *data, size_t len) {
-    if (_delay_close) {
-        // 正在延时关闭中，忽略所有数据
-        return;
-    }
     if (!isRtp(data, len)) {
         // 忽略非rtp数据
         WarnP(this) << "Not rtp packet";
@@ -104,33 +100,31 @@ void RtpSession::onRtpPacket(const char *data, size_t len) {
             return;
         }
     }
-    if (!_process) {
+
-        //未设置ssrc时，尝试获取ssrc
+    // 未设置ssrc时，尝试获取ssrc
-        if (!_ssrc && !RtpSelector::getSSRC(data, len, _ssrc)) {
+    if (!_ssrc && !getSSRC(data, len, _ssrc)) {
         return;
     }
+
+    // 未指定流id就使用ssrc为流id
     if (_stream_id.empty()) {
-            //未指定流id就使用ssrc为流id
         _stream_id = printSSRC(_ssrc);
     }
-        try {
+
-            _process = RtpSelector::Instance().getProcess(_stream_id, true);
+    if (!_process) {
-        } catch (RtpSelector::ProcessExisted &ex) {
+        _process = RtpProcess::createProcess(_stream_id);
-            if (!_is_udp) {
-                // tcp情况下立即断开连接
-                throw;
-            }
-            // udp情况下延时断开连接(等待超时自动关闭)，防止频繁创建销毁RtpSession对象
-            WarnP(this) << ex.what();
-            _delay_close = true;
-            return;
-        }
         _process->setOnlyTrack((RtpProcess::OnlyTrack)_only_track);
-        _process->setDelegate(static_pointer_cast<RtpSession>(shared_from_this()));
+        weak_ptr<RtpSession>  weak_self = static_pointer_cast<RtpSession>(shared_from_this());
+        _process->setOnDetach([weak_self](const SockException &ex) {
+            if (auto strong_self = weak_self.lock()) {
+                strong_self->_process = nullptr;
+                strong_self->shutdown(ex);
+            }
+        });
     }
     try {
         uint32_t rtp_ssrc = 0;
-        RtpSelector::getSSRC(data, len, rtp_ssrc);
+        getSSRC(data, len, rtp_ssrc);
         if (rtp_ssrc != _ssrc) {
             WarnP(this) << "ssrc mismatched, rtp dropped: " << rtp_ssrc << " != " << _ssrc;
             return;
@@ -143,26 +137,10 @@ void RtpSession::onRtpPacket(const char *data, size_t len) {
         } else {
             throw;
         }
-    } catch (std::exception &ex) {
-        if (!_is_udp) {
-            // tcp情况下立即断开连接
-            throw;
-        }
-        // udp情况下延时断开连接(等待超时自动关闭)，防止频繁创建销毁RtpSession对象
-        WarnP(this) << ex.what();
-        _delay_close = true;
-        return;
     }
     _ticker.resetTime();
 }
 
-bool RtpSession::close(MediaSource &sender) {
-    //此回调在其他线程触发
-    string err = StrPrinter << "close media: " << sender.getUrl();
-    safeShutdown(SockException(Err_shutdown, err));
-    return true;
-}
-
 static const char *findSSRC(const char *data, ssize_t len, uint32_t ssrc) {
     // rtp前面必须预留两个字节的长度字段
     for (ssize_t i = 2; i <= len - 4; ++i) {
@@ -268,7 +246,7 @@ const char *RtpSession::searchByPsHeaderFlag(const char *data, size_t len) {
 
     // TODO or Not ? 更新设置ssrc
     uint32_t rtp_ssrc = 0;
-    RtpSelector::getSSRC(rtp_ptr + 2, len, rtp_ssrc);
+    getSSRC(rtp_ptr + 2, len, rtp_ssrc);
     _ssrc = rtp_ssrc;
     InfoL << "设置_ssrc为：" << _ssrc;
     // RtpServer::updateSSRC(uint32_t ssrc)

src/Rtp/RtpSession.h

@@ -20,7 +20,7 @@
 
 namespace mediakit{
 
-class RtpSession : public toolkit::Session, public RtpSplitter, public MediaSourceEvent {
+class RtpSession : public toolkit::Session, public RtpSplitter {
 public:
     static const std::string kStreamID;
     static const std::string kSSRC;
@@ -34,10 +34,9 @@ public:
     void onManager() override;
     void setParams(toolkit::mINI &ini);
     void attachServer(const toolkit::Server &server) override;
+    void setRtpProcess(RtpProcess::Ptr process);
 
 protected:
-    // 通知其停止推流
-    bool close(MediaSource &sender) override;
     // 收到rtp回调
     void onRtpPacket(const char *data, size_t len) override;
     // RtpSplitter override
@@ -48,10 +47,10 @@ protected:
     const char *searchByPsHeaderFlag(const char *data, size_t len);
 
 private:
-    bool _delay_close = false;
     bool _is_udp = false;
     bool _search_rtp = false;
     bool _search_rtp_finished = false;
+    bool _emit_detach = false;
     int _only_track = 0;
     uint32_t _ssrc = 0;
     toolkit::Ticker _ticker;

src/Rtsp/RtpMultiCaster.cpp

@@ -144,7 +144,16 @@ RtpMultiCaster::RtpMultiCaster(SocketHelper &helper, const string &local_ip, con
         });
     });
 
-    _rtp_reader->setDetachCB([this]() {
+    string strKey = StrPrinter << local_ip << " " << vhost << " " << app << " " << stream << endl;
+    _rtp_reader->setDetachCB([this, strKey]() {
+        {
+            lock_guard<recursive_mutex> lck(g_mtx);
+            auto it = g_multi_caster_map.find(strKey);
+            if (it != g_multi_caster_map.end()) {
+                g_multi_caster_map.erase(it);
+            }
+        }
+
         unordered_map<void *, onDetach> _detach_map_copy;
         {
             lock_guard<recursive_mutex> lck(_mtx);

src/Rtsp/Rtsp.cpp

@@ -470,6 +470,15 @@ string printSSRC(uint32_t ui32Ssrc) {
     return tmp;
 }
 
+bool getSSRC(const char *data, size_t data_len, uint32_t &ssrc) {
+    if (data_len < 12) {
+        return false;
+    }
+    uint32_t *ssrc_ptr = (uint32_t *)(data + 8);
+    ssrc = ntohl(*ssrc_ptr);
+    return true;
+}
+
 bool isRtp(const char *buf, size_t size) {
     if (size < 2) {
         return false;

src/Rtsp/Rtsp.h

@@ -317,6 +317,7 @@ toolkit::Buffer::Ptr makeRtpOverTcpPrefix(uint16_t size, uint8_t interleaved);
 void makeSockPair(std::pair<toolkit::Socket::Ptr, toolkit::Socket::Ptr> &pair, const std::string &local_ip, bool re_use_port = false, bool is_udp = true);
 // 十六进制方式打印ssrc
 std::string printSSRC(uint32_t ui32Ssrc);
+bool getSSRC(const char *data, size_t data_len, uint32_t &ssrc);
 
 bool isRtp(const char *buf, size_t size);
 bool isRtcp(const char *buf, size_t size);

src/Rtsp/RtspPlayer.cpp

@@ -28,6 +28,7 @@ using namespace std;
 namespace mediakit {
 
 enum PlayType { type_play = 0, type_pause, type_seek, type_speed };
+enum class BeatType : uint32_t { both = 0, rtcp, cmd  };
 
 RtspPlayer::RtspPlayer(const EventPoller::Ptr &poller)
     : TcpClient(poller) {}
@@ -85,6 +86,8 @@ void RtspPlayer::play(const string &strUrl) {
 
     _play_url = url._url;
     _rtp_type = (Rtsp::eRtpType)(int)(*this)[Client::kRtpType];
+    _beat_type = (*this)[Client::kRtspBeatType].as<int>();
+    _beat_interval_ms = (*this)[Client::kBeatIntervalMS].as<int>();
     DebugL << url._url << " " << (url._user.size() ? url._user : "null") << " " << (url._passwd.size() ? url._passwd : "null") << " " << _rtp_type;
 
     weak_ptr<RtspPlayer> weakSelf = static_pointer_cast<RtspPlayer>(shared_from_this());
@@ -210,7 +213,8 @@ void RtspPlayer::handleResDESCRIBE(const Parser &parser) {
     if (play_track != TrackInvalid) {
         auto track = sdpParser.getTrack(play_track);
         _sdp_track.emplace_back(track);
-        sdp = track->toString();
+        auto title_track = sdpParser.getTrack(TrackTitle);
+        sdp = (title_track ? title_track->toString() : "") + track->toString();
     } else {
         _sdp_track = sdpParser.getAvailableTrack();
         sdp = sdpParser.toString();
@@ -641,23 +645,28 @@ void RtspPlayer::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_idx) {
     rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 
     auto &ticker = _rtcp_send_ticker[track_idx];
-    if (ticker.elapsedTime() < 3 * 1000) {
+    if (ticker.elapsedTime() < _beat_interval_ms) {
-        // 时间未到
+        // 心跳时间未到
         return;
     }
-    auto &rtcp_flag = _send_rtcp[track_idx];
 
-    // 每3秒发送一次心跳，rtcp与rtsp信令轮流心跳，该特性用于兼容issue:642
+    // 有些rtsp服务器需要rtcp保活，有些需要发送信令保活; rtcp与rtsp信令轮流心跳，该特性用于兼容issue:#642
-    // 有些rtsp服务器需要rtcp保活，有些需要发送信令保活
+    auto &rtcp_flag = _send_rtcp[track_idx];
+    ticker.resetTime();
+
+    switch ((BeatType)_beat_type) {
+        case BeatType::cmd: rtcp_flag = false; break;
+        case BeatType::rtcp: rtcp_flag = true; break;
+        case BeatType::both:
+        default: rtcp_flag = !rtcp_flag; break;
+    }
 
     // 发送信令保活
     if (!rtcp_flag) {
         if (track_idx == 0) {
+            // 两个track无需同时触发发送信令保活
             sendKeepAlive();
         }
-        ticker.resetTime();
-        // 下次发送rtcp保活
-        rtcp_flag = true;
         return;
     }
 
@@ -679,9 +688,6 @@ void RtspPlayer::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_idx) {
     rtcp_sdes->chunks.ssrc = htonl(ssrc);
     send_rtcp(this, track_idx, std::move(rtcp));
     send_rtcp(this, track_idx, RtcpHeader::toBuffer(rtcp_sdes));
-    ticker.resetTime();
-    // 下次发送信令保活
-    rtcp_flag = false;
 }
 
 void RtspPlayer::onPlayResult_l(const SockException &ex, bool handshake_done) {

src/Rtsp/RtspPlayer.h

@@ -114,6 +114,11 @@ private:
     //轮流发送rtcp与GET_PARAMETER保活
     bool _send_rtcp[2] = {true, true};
 
+    // 心跳类型
+    uint32_t _beat_type = 0;
+    // 心跳保护间隔
+    uint32_t _beat_interval_ms = 0;
+
     std::string _play_url;
     std::vector<SdpTrack::Ptr> _sdp_track;
     std::function<void(const Parser&)> _on_response;

src/Rtsp/RtspSplitter.cpp

@@ -10,8 +10,10 @@
 
 #include <cstdlib>
 #include "RtspSplitter.h"
-#include "Util/logger.h"
 #include "Util/util.h"
+#include "Util/logger.h"
+#include "Common/macros.h"
+#include "Rtsp/RtpReceiver.h"
 
 using namespace std;
 using namespace toolkit;
@@ -58,13 +60,28 @@ const char *RtspSplitter::onSearchPacketTail_l(const char *data, size_t len) {
 
 ssize_t RtspSplitter::onRecvHeader(const char *data, size_t len) {
     if (_isRtpPacket) {
+        try {
             onRtpPacket(data, len);
+        } catch (RtpTrack::BadRtpException &ex) {
+            WarnL << ex.what();
+        }
         return 0;
     }
     if (len == 4 && !memcmp(data, "\r\n\r\n", 4)) {
         return 0;
     }
+    try {
         _parser.parse(data, len);
+    } catch (mediakit::AssertFailedException &ex){
+        if (!_enableRecvRtp) {
+            // 还在握手中，直接中断握手
+            throw;
+        }
+        // 握手已经结束，如果rtsp server存在发送缓存溢出的bug，那么rtsp信令可能跟rtp混在一起
+        // 这种情况下，rtsp信令解析异常不中断链接，只丢弃这个包
+        WarnL << ex.what();
+        return 0;
+    }
     auto ret = getContentLength(_parser);
     if (ret == 0) {
         onWholeRtspPacket(_parser);

tests/test_bench_push.cpp

@@ -20,7 +20,7 @@
 #include "Thread/WorkThreadPool.h"
 #include "Pusher/MediaPusher.h"
 #include "Player/PlayerProxy.h"
-
+#include "Record/MP4Reader.h"
 using namespace std;
 using namespace toolkit;
 using namespace mediakit;
@@ -52,7 +52,7 @@ public:
                              Option::ArgRequired,/*该选项后面必须跟值*/
                              nullptr,/*该选项默认值*/
                              true,/*该选项是否必须赋值，如果没有默认值且为ArgRequired时用户必须提供该参数否则将抛异常*/
-                             "拉流url,支持rtsp/rtmp/hls",/*该选项说明文字*/
+                             "拉流url,支持rtsp/rtmp/hls/mp4文件",/*该选项说明文字*/
                              nullptr);
 
         (*_parser) << Option('o',/*该选项简称，如果是\x00则说明无简称*/
@@ -92,18 +92,16 @@ public:
                              Option::ArgRequired,/*该选项后面必须跟值*/
                              to_string((int) (Rtsp::RTP_TCP)).data(),/*该选项默认值*/
                              true,/*该选项是否必须赋值，如果没有默认值且为ArgRequired时用户必须提供该参数否则将抛异常*/
-                             "rtsp拉流和推流方式,支持tcp/udp:0/1",/*该选项说明文字*/
+                            "rtsp拉流和推流方式,支持tcp/udp:0/1", /*该选项说明文字*/
                             nullptr);
     }
 
     ~CMD_main() override {}
 
-    const char *description() const override {
+    const char *description() const override { return "主程序命令参数"; }
-        return "主程序命令参数";
-    }
 };
 
-//此程序用于推流性能测试
+// 此程序用于推流性能测试
 int main(int argc, char *argv[]) {
     CMD_main cmd_main;
     try {
@@ -116,7 +114,7 @@ int main(int argc, char *argv[]) {
     }
 
     int threads = cmd_main["threads"];
-    LogLevel logLevel = (LogLevel) cmd_main["level"].as<int>();
+    LogLevel logLevel = (LogLevel)cmd_main["level"].as<int>();
     logLevel = MIN(MAX(logLevel, LTrace), LError);
     auto in_url = cmd_main["in"];
     auto out_url = cmd_main["out"];
@@ -129,6 +127,8 @@ int main(int argc, char *argv[]) {
         cout << "推流协议只支持rtsp或rtmp！" << endl;
         return -1;
     }
+    const std::string app = "app";
+    const std::string stream = "test";
 
     //设置日志
     Logger::Instance().add(std::make_shared<ConsoleChannel>("ConsoleChannel", logLevel));
@@ -145,22 +145,39 @@ int main(int argc, char *argv[]) {
     ProtocolOption option;
     option.enable_hls = false;
     option.enable_mp4 = false;
+    MediaSource::Ptr src = nullptr;
+    PlayerProxy::Ptr proxy = nullptr;;
 
+    if (end_with(in_url, ".mp4")) {
+        // create MediaSource from mp4file
+        auto reader = std::make_shared<MP4Reader>(DEFAULT_VHOST, app, stream, in_url);
+        //mp4 repeat
+        reader->startReadMP4(0, true, true);
+        src = MediaSource::find(schema, DEFAULT_VHOST, app, stream, false);
+        if (!src) {
+            // mp4文件不存在
+            WarnL << "no such file or directory: " << in_url;
+            return -1;
+        }
+    } else {
         //添加拉流代理
-    auto proxy = std::make_shared<PlayerProxy>(DEFAULT_VHOST, "app", "test", option);
+        proxy = std::make_shared<PlayerProxy>(DEFAULT_VHOST, app, stream, option);
         //rtsp拉流代理方式
         (*proxy)[Client::kRtpType] = rtp_type;
         //开始拉流代理
         proxy->play(in_url);
 
-    auto get_src = [schema]() {
+    }
-        return MediaSource::find(schema, DEFAULT_VHOST, "app", "test", false);
+
+    auto get_src = [schema,app,stream]() {
+        return MediaSource::find(schema, DEFAULT_VHOST, app, stream, false);
     };
 
     //推流器map
     recursive_mutex mtx;
     unordered_map<void *, MediaPusher::Ptr> pusher_map;
 
+
     auto add_pusher = [&](const MediaSource::Ptr &src, const string &rand_str, size_t index) {
         auto pusher = std::make_shared<MediaPusher>(src);
         auto tag = pusher.get();

tests/test_rtp.cpp

@@ -17,7 +17,7 @@
 #include "Rtsp/RtspSession.h"
 #include "Rtmp/RtmpSession.h"
 #include "Http/HttpSession.h"
-#include "Rtp/RtpSelector.h"
+#include "Rtp/RtpProcess.h"
 
 using namespace std;
 using namespace toolkit;
@@ -42,7 +42,7 @@ static bool loadFile(const char *path, const EventPoller::Ptr &poller) {
     memset(&addr, 0, sizeof(addr));
     addr.ss_family = AF_INET;
     auto sock = Socket::createSocket(poller);
-    auto process = RtpSelector::Instance().getProcess("test", true);
+    auto process = RtpProcess::createProcess("test");
 
     uint64_t stamp_last = 0;
     auto total_size = std::make_shared<size_t>(0);
@@ -89,7 +89,6 @@ static bool loadFile(const char *path, const EventPoller::Ptr &poller) {
         auto ret = do_read();
         if (!ret) {
             WarnL << *total_size / 1024 << "KB";
-            RtpSelector::Instance().delProcess("test", process.get());
         }
         return ret;
     });

webrtc/DtlsTransport.cpp

@@ -383,7 +383,7 @@ namespace RTC
         });
         // Set ciphers.
         ret = SSL_CTX_set_cipher_list(
-          sslCtx, "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK");
+          sslCtx, "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK:!RC4");
 
         if (ret == 0)
         {

webrtc/Nack.cpp

@@ -9,24 +9,56 @@
  */
 
 #include "Nack.h"
+#include "Common/config.h"
 
 using namespace std;
 using namespace toolkit;
 
 namespace mediakit {
 
-static constexpr uint32_t kMaxNackMS = 5 * 1000;
+// RTC配置项目
-static constexpr uint32_t kRtpCacheCheckInterval = 100;
+namespace Rtc {
+#define RTC_FIELD "rtc."
+//~ nack接收端
+// Nack缓存包最早时间间隔
+const string kMaxNackMS = RTC_FIELD "maxNackMS";
+// Nack包检查间隔(包数量)
+const string kRtpCacheCheckInterval = RTC_FIELD "rtpCacheCheckInterval";
+//~ nack发送端
+//最大保留的rtp丢包状态个数
+const string kNackMaxSize = RTC_FIELD "nackMaxSize";
+// rtp丢包状态最长保留时间
+const string kNackMaxMS = RTC_FIELD "nackMaxMS";
+// nack最多请求重传次数
+const string kNackMaxCount = RTC_FIELD "nackMaxCount";
+// nack重传频率，rtt的倍数
+const string kNackIntervalRatio = RTC_FIELD "nackIntervalRatio";
+// nack包中rtp个数，减小此值可以让nack包响应更灵敏
+const string kNackRtpSize = RTC_FIELD "nackRtpSize";
+
+static onceToken token([]() {
+    mINI::Instance()[kMaxNackMS] = 5 * 1000;
+    mINI::Instance()[kRtpCacheCheckInterval] = 100;
+    mINI::Instance()[kNackMaxSize] = 2048;
+    mINI::Instance()[kNackMaxMS] = 3 * 1000;
+    mINI::Instance()[kNackMaxCount] = 15;
+    mINI::Instance()[kNackIntervalRatio] = 1.0f;
+    mINI::Instance()[kNackRtpSize] = 8;
+});
+
+} // namespace Rtc
 
 void NackList::pushBack(RtpPacket::Ptr rtp) {
     auto seq = rtp->getSeq();
     _nack_cache_seq.emplace_back(seq);
     _nack_cache_pkt.emplace(seq, std::move(rtp));
-    if (++_cache_ms_check < kRtpCacheCheckInterval) {
+    GET_CONFIG(uint32_t, rtpcache_checkinterval, Rtc::kRtpCacheCheckInterval);
+    if (++_cache_ms_check < rtpcache_checkinterval) {
         return;
     }
     _cache_ms_check = 0;
-    while (getCacheMS() >= kMaxNackMS) {
+    GET_CONFIG(uint32_t, maxnackms, Rtc::kMaxNackMS);
+    while (getCacheMS() >= maxnackms) {
         // 需要清除部分nack缓存
         popFront();
     }
@@ -148,10 +180,13 @@ void NackContext::makeNack(uint16_t max_seq, bool flush) {
     eraseFrontSeq();
     // 最多生成5个nack包，防止seq大幅跳跃导致一直循环
     auto max_nack = 5u;
+    GET_CONFIG(uint32_t, nack_rtpsize, Rtc::kNackRtpSize);
+    // kNackRtpSize must between 0 and 16
+    nack_rtpsize = std::min<uint32_t>(nack_rtpsize, FCI_NACK::kBitSize);
     while (_nack_seq != max_seq && max_nack--) {
         // 一次不能发送超过16+1个rtp的状态
         uint16_t nack_rtp_count = std::min<uint16_t>(FCI_NACK::kBitSize, max_seq - (uint16_t)(_nack_seq + 1));
-        if (!flush && nack_rtp_count < kNackRtpSize) {
+        if (!flush && nack_rtp_count < nack_rtpsize) {
             // 非flush状态下，seq个数不足以发送一次nack
             break;
         }
@@ -206,7 +241,9 @@ void NackContext::clearNackStatus(uint16_t seq) {
     _nack_send_status.erase(it);
 
     // 限定rtt在合理有效范围内
-    _rtt = max<int>(10, min<int>(rtt, kNackMaxMS / kNackMaxCount));
+    GET_CONFIG(uint32_t, nack_maxms, Rtc::kNackMaxMS);
+    GET_CONFIG(uint32_t, nack_maxcount, Rtc::kNackMaxCount);
+    _rtt = max<int>(10, min<int>(rtt, nack_maxms / nack_maxcount));
 }
 
 void NackContext::recordNack(const FCI_NACK &nack) {
@@ -222,7 +259,8 @@ void NackContext::recordNack(const FCI_NACK &nack) {
         ++i;
     }
     // 记录太多了，移除一部分早期的记录
-    while (_nack_send_status.size() > kNackMaxSize) {
+    GET_CONFIG(uint32_t, nack_maxsize, Rtc::kNackMaxSize);
+    while (_nack_send_status.size() > nack_maxsize) {
         _nack_send_status.erase(_nack_send_status.begin());
     }
 }
@@ -230,13 +268,16 @@ void NackContext::recordNack(const FCI_NACK &nack) {
 uint64_t NackContext::reSendNack() {
     set<uint16_t> nack_rtp;
     auto now = getCurrentMillisecond();
+    GET_CONFIG(uint32_t, nack_maxms, Rtc::kNackMaxMS);
+    GET_CONFIG(uint32_t, nack_maxcount, Rtc::kNackMaxCount);
+    GET_CONFIG(float, nack_intervalratio, Rtc::kNackIntervalRatio);
     for (auto it = _nack_send_status.begin(); it != _nack_send_status.end();) {
-        if (now - it->second.first_stamp > kNackMaxMS) {
+        if (now - it->second.first_stamp > nack_maxms) {
             // 该rtp丢失太久了，不再要求重传
             it = _nack_send_status.erase(it);
             continue;
         }
-        if (now - it->second.update_stamp < kNackIntervalRatio * _rtt) {
+        if (now - it->second.update_stamp < nack_intervalratio * _rtt) {
             // 距离上次nack不足2倍的rtt，不用再发送nack
             ++it;
             continue;
@@ -245,7 +286,7 @@ uint64_t NackContext::reSendNack() {
         nack_rtp.emplace(it->first);
         // 更新nack发送时间戳
         it->second.update_stamp = now;
-        if (++(it->second.nack_count) == kNackMaxCount) {
+        if (++(it->second.nack_count) == nack_maxcount) {
             // nack次数太多，移除之
             it = _nack_send_status.erase(it);
             continue;

webrtc/Nack.h

@@ -41,18 +41,6 @@ class NackContext {
 public:
     using Ptr = std::shared_ptr<NackContext>;
     using onNack = std::function<void(const FCI_NACK &nack)>;
-    //最大保留的rtp丢包状态个数
-    static constexpr auto kNackMaxSize = 2048;
-    // rtp丢包状态最长保留时间
-    static constexpr auto kNackMaxMS = 3 * 1000;
-    // nack最多请求重传10次
-    static constexpr auto kNackMaxCount = 15;
-    // nack重传频率，rtt的倍数
-    static constexpr auto kNackIntervalRatio = 1.0f;
-    // nack包中rtp个数，减小此值可以让nack包响应更灵敏
-    static constexpr auto kNackRtpSize = 8;
-
-    static_assert(kNackRtpSize >=0 && kNackRtpSize <= FCI_NACK::kBitSize, "NackContext::kNackRtpSize must between 0 and 16");
 
     NackContext();
 

webrtc/WebRtcTransport.cpp

@@ -57,6 +57,9 @@ const string kMinBitrate = RTC_FIELD "min_bitrate";
 // 数据通道设置
 const string kDataChannelEcho = RTC_FIELD "datachannel_echo";
 
+// rtp丢包状态最长保留时间
+const string kNackMaxMS = RTC_FIELD "nackMaxMS";
+
 static onceToken token([]() {
     mINI::Instance()[kTimeOutSec] = 15;
     mINI::Instance()[kExternIP] = "";
@@ -69,6 +72,8 @@ static onceToken token([]() {
     mINI::Instance()[kMinBitrate] = 0;
 
     mINI::Instance()[kDataChannelEcho] = true;
+
+    mINI::Instance()[kNackMaxMS] = 3 * 1000;
 });
 
 } // namespace RTC
@@ -800,7 +805,8 @@ public:
         _on_nack = std::move(on_nack);
         setOnSorted(std::move(cb));
         //设置jitter buffer参数
-        RtpTrackImp::setParams(1024, NackContext::kNackMaxMS, 512);
+        GET_CONFIG(uint32_t, nack_maxms, Rtc::kNackMaxMS);
+        RtpTrackImp::setParams(1024, nack_maxms, 512);
         _nack_ctx.setOnNack([this](const FCI_NACK &nack) { onNack(nack); });
     }
 

webrtc_player/android/.gitignore

@@ -1,16 +0,0 @@
-*.iml
-.gradle
-/local.properties
-/.idea/caches
-/.idea/libraries
-/.idea/modules.xml
-/.idea/workspace.xml
-/.idea/navEditor.xml
-/.idea/assetWizardSettings.xml
-.DS_Store
-/build
-/captures
-.externalNativeBuild
-.cxx
-local.properties
-/.idea/

webrtc_player/android/app/.gitignore

@@ -1,2 +0,0 @@
-/build
-.cxx

webrtc_player/android/app/build.gradle

@@ -1,54 +0,0 @@
-plugins {
-    id 'com.android.application'
-    id 'org.jetbrains.kotlin.android'
-    id 'kotlin-android-extensions'
-    id 'kotlin-kapt'
-
-}
-apply plugin: 'kotlin-android'
-
-android {
-    compileSdk 32
-
-    defaultConfig {
-        applicationId "com.zlmediakit.webrtc"
-        minSdk 21
-        targetSdk 32
-        versionCode 1
-        versionName "1.0"
-
-        testInstrumentationRunner "androidx.test.runner.AndroidJUnitRunner"
-    }
-
-    buildTypes {
-        release {
-            minifyEnabled false
-            proguardFiles getDefaultProguardFile('proguard-android-optimize.txt'), 'proguard-rules.pro'
-        }
-    }
-    compileOptions {
-        sourceCompatibility JavaVersion.VERSION_1_8
-        targetCompatibility JavaVersion.VERSION_1_8
-    }
-    kotlinOptions {
-        jvmTarget = '1.8'
-    }
-}
-
-dependencies {
-
-    implementation 'androidx.core:core-ktx:1.7.0'
-    implementation 'androidx.appcompat:appcompat:1.5.1'
-    implementation 'com.google.android.material:material:1.6.1'
-    implementation 'androidx.constraintlayout:constraintlayout:2.1.4'
-    testImplementation 'junit:junit:4.13.2'
-    androidTestImplementation 'androidx.test.ext:junit:1.1.3'
-    androidTestImplementation 'androidx.test.espresso:espresso-core:3.4.0'
-    implementation 'com.google.code.gson:gson:2.8.9'
-
-    implementation("com.squareup.okhttp3:okhttp:4.10.0")
-    implementation "org.jetbrains.kotlin:kotlin-stdlib-jdk7:$kotlin_version"
-
-    implementation 'org.webrtc:google-webrtc:1.0.32006'
-
-}

webrtc_player/android/app/proguard-rules.pro

@@ -1,21 +0,0 @@
-# Add project specific ProGuard rules here.
-# You can control the set of applied configuration files using the
-# proguardFiles setting in build.gradle.
-#
-# For more details, see
-#   http://developer.android.com/guide/developing/tools/proguard.html
-
-# If your project uses WebView with JS, uncomment the following
-# and specify the fully qualified class name to the JavaScript interface
-# class:
-#-keepclassmembers class fqcn.of.javascript.interface.for.webview {
-#   public *;
-#}
-
-# Uncomment this to preserve the line number information for
-# debugging stack traces.
-#-keepattributes SourceFile,LineNumberTable
-
-# If you keep the line number information, uncomment this to
-# hide the original source file name.
-#-renamesourcefileattribute SourceFile

webrtc_player/android/app/src/androidTest/java/com/zlmediakit/webrtc/ExampleInstrumentedTest.kt

@@ -1,24 +0,0 @@
-package com.zlmediakit.webrtc
-
-import androidx.test.platform.app.InstrumentationRegistry
-import androidx.test.ext.junit.runners.AndroidJUnit4
-
-import org.junit.Test
-import org.junit.runner.RunWith
-
-import org.junit.Assert.*
-
-/**
- * Instrumented test, which will execute on an Android device.
- *
- * See [testing documentation](http://d.android.com/tools/testing).
- */
-@RunWith(AndroidJUnit4::class)
-class ExampleInstrumentedTest {
-    @Test
-    fun useAppContext() {
-        // Context of the app under test.
-        val appContext = InstrumentationRegistry.getInstrumentation().targetContext
-        assertEquals("com.zlmediakit.webrtc", appContext.packageName)
-    }
-}

webrtc_player/android/app/src/main/AndroidManifest.xml

@@ -1,46 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
-    xmlns:tools="http://schemas.android.com/tools"
-    package="com.zlmediakit.webrtc">
-
-    <uses-feature android:name="android.hardware.camera"/>
-    <uses-feature android:name="android.hardware.camera.autofocus"/>
-    <uses-feature
-        android:glEsVersion="0x00020000"
-        android:required="true"/>
-
-    <uses-permission android:name="android.permission.CAMERA"/>
-    <uses-permission android:name="android.permission.CHANGE_NETWORK_STATE"/>
-    <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS"/>
-    <uses-permission android:name="android.permission.RECORD_AUDIO"/>
-    <uses-permission android:name="android.permission.INTERNET"/>
-    <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
-    <uses-permission android:name="android.permission.READ_EXTERNAL_STORAGE"/>
-    <uses-permission android:name="android.permission.ACCESS_NETWORK_STATE"/>
-    <uses-permission android:name="android.permission.CAPTURE_VIDEO_OUTPUT"/>
-    <uses-permission android:name="android.permission.READ_PHONE_STATE"/>
-    <application
-        android:allowBackup="true"
-        android:dataExtractionRules="@xml/data_extraction_rules"
-        android:fullBackupContent="@xml/backup_rules"
-        android:icon="@mipmap/ic_launcher"
-        android:label="@string/app_name"
-        android:roundIcon="@mipmap/ic_launcher_round"
-        android:supportsRtl="true"
-        android:theme="@style/Theme.AndroidWebRTC"
-        android:usesCleartextTraffic="true"
-
-
-        tools:targetApi="31">
-        <activity
-            android:name=".MainActivity"
-            android:exported="true">
-            <intent-filter>
-                <action android:name="android.intent.action.MAIN" />
-
-                <category android:name="android.intent.category.LAUNCHER" />
-            </intent-filter>
-        </activity>
-    </application>
-
-</manifest>

webrtc_player/android/app/src/main/java/com/zlmediakit/webrtc/MainActivity.kt

@@ -1,79 +0,0 @@
-package com.zlmediakit.webrtc
-
-import android.annotation.SuppressLint
-import android.graphics.drawable.BitmapDrawable
-import android.graphics.drawable.Drawable
-import android.os.Bundle
-import android.widget.Toast
-import androidx.appcompat.app.AppCompatActivity
-import kotlinx.android.synthetic.main.activity_main.*
-import kotlinx.android.synthetic.main.activity_main.view.*
-
-
-class MainActivity : AppCompatActivity() {
-
-    private var isSpeaker = true
-
-    @SuppressLint("SetTextI18n")
-    override fun onCreate(savedInstanceState: Bundle?) {
-        super.onCreate(savedInstanceState)
-        setContentView(R.layout.activity_main)
-
-        lifecycle.addObserver(web_rtc_sv)
-
-        //http://124.223.98.45/index/api/webrtc?app=live&stream=test&type=play
-        url.setText("http://124.223.98.45/index/api/webrtc?app=live&stream=test&type=play")
-
-        //http://192.168.1.17/index/api/webrtc?app=live&stream=test&type=play
-        btn_play.setOnClickListener {
-            web_rtc_sv?.setVideoPath(url.text.toString())
-            web_rtc_sv.start()
-        }
-
-        web_rtc_sv.setOnErrorListener { errorCode, errorMsg ->
-            runOnUiThread {
-                Toast.makeText(this, "errorCode:$errorCode,errorMsg:$errorMsg", Toast.LENGTH_SHORT)
-                    .show()
-            }
-        }
-
-
-        btn_pause.setOnClickListener {
-            web_rtc_sv?.pause()
-        }
-
-        btn_resume.setOnClickListener {
-            web_rtc_sv?.resume()
-        }
-
-        btn_screenshot.setOnClickListener {
-            web_rtc_sv?.screenshot {
-                runOnUiThread {
-                    iv_screen.setImageDrawable(BitmapDrawable(it))
-                }
-            }
-        }
-
-        btn_mute.setOnClickListener {
-            web_rtc_sv.mute(true)
-        }
-
-
-        selectAudio()
-        btn_speaker.setOnClickListener {
-            selectAudio()
-        }
-
-    }
-
-    fun selectAudio(){
-        if (isSpeaker){
-            btn_speaker.setText("扬声器")
-            web_rtc_sv.setSpeakerphoneOn(isSpeaker)
-        }else{
-            btn_speaker.setText("话筒")
-            web_rtc_sv.setSpeakerphoneOn(isSpeaker)
-        }
-        isSpeaker=!isSpeaker
-    }
-}

webrtc_player/android/app/src/main/java/com/zlmediakit/webrtc/WebRTCSurfaceView.kt

@@ -1,439 +0,0 @@
-package com.zlmediakit.webrtc
-
-import android.content.Context
-import android.graphics.Bitmap
-import android.media.AudioManager
-import android.util.AttributeSet
-import android.util.Log
-import android.view.LayoutInflater
-import android.widget.RelativeLayout
-import androidx.lifecycle.DefaultLifecycleObserver
-import androidx.lifecycle.LifecycleOwner
-import com.google.gson.Gson
-import okhttp3.*
-import okhttp3.MediaType.Companion.toMediaType
-import okhttp3.MediaType.Companion.toMediaTypeOrNull
-import org.webrtc.*
-import org.webrtc.RendererCommon.ScalingType
-import org.webrtc.audio.AudioDeviceModule
-import org.webrtc.audio.JavaAudioDeviceModule
-import java.io.IOException
-import java.util.*
-
-public class WebRTCSurfaceView(context: Context, attrs: AttributeSet?) :
-    RelativeLayout(context, attrs), DefaultLifecycleObserver, RendererCommon.RendererEvents {
-
-
-    private data class sdp(var sdp: String, var username: String, var password: String)
-
-    private data class SdpResponse(var code: Int, var id: String, var sdp: String, var type: String)
-
-    private enum class ErrorCode(val errorCode: Int) {
-        SUCCESS(0x00),
-        GET_REMOTE_SDP_ERROR(0x01);
-    }
-
-
-    companion object {
-        private val TAG = "WebRTCSurfaceView"
-
-    }
-
-    private var mContext: Context = context
-
-    private val eglBase: EglBase = EglBase.create()
-    private var mEGLBaseContext: EglBase.Context = eglBase.eglBaseContext
-
-    private lateinit var videoUrl: String;
-
-    private var mPeerConnectionFactory: PeerConnectionFactory? = null
-
-    private var mLocalMediaStream: MediaStream? = null
-    private var mLocalAudioTrack: AudioTrack? = null
-    private var mAudioSource: AudioSource? = null
-
-    private var mLocalSessionDescription: SessionDescription? = null
-    private var mRemoteSessionDescription: SessionDescription? = null
-
-    private var mLocalPeer: Peer? = null
-
-    private var mSurfaceViewRenderer: SurfaceViewRenderer
-
-    private lateinit var OnErrorListener: (errorCode: Int, errorMsg: String) -> Unit?
-
-    fun setOnErrorListener(listener: (errorCode: Int, errorMsg: String) -> Unit) {
-        this.OnErrorListener = listener
-    }
-
-    private lateinit var OnPreparedListener: () -> Unit?
-
-    fun setOnPreparedListener(listener: () -> Unit) {
-        this.OnPreparedListener = listener
-    }
-
-    private val audioManager: AudioManager
-
-
-    init {
-
-        val view = LayoutInflater.from(mContext).inflate(R.layout.layout_videoview, this)
-
-        mPeerConnectionFactory = createConnectionFactory()
-
-        mSurfaceViewRenderer = view.findViewById(R.id.surface_view_renderer)
-
-        mSurfaceViewRenderer.init(mEGLBaseContext, this)
-        mSurfaceViewRenderer.setScalingType(ScalingType.SCALE_ASPECT_FILL)
-        mSurfaceViewRenderer.setEnableHardwareScaler(true)
-
-
-        //创建媒体流
-        mLocalMediaStream = mPeerConnectionFactory?.createLocalMediaStream("ARDAMS")
-        //采集音频
-        mAudioSource = mPeerConnectionFactory?.createAudioSource(createAudioConstraints())
-        mLocalAudioTrack = mPeerConnectionFactory?.createAudioTrack("ARDAMSa0", mAudioSource)
-
-        //添加Tracks
-        mLocalMediaStream?.addTrack(mLocalAudioTrack)
-
-        audioManager = context.getSystemService(Context.AUDIO_SERVICE) as AudioManager
-        audioManager.isSpeakerphoneOn = false
-
-
-    }
-
-
-    private fun set(width: Int, height: Int) {
-        layoutParams.width = width
-        layoutParams.height = height
-    }
-
-    private fun createConnectionFactory(): PeerConnectionFactory? {
-
-        val options = PeerConnectionFactory.InitializationOptions.builder(mContext)
-            .setEnableInternalTracer(false)
-            .createInitializationOptions()
-
-        PeerConnectionFactory.initialize(options)
-
-        val videoEncoderFactory = DefaultVideoEncoderFactory(
-            mEGLBaseContext,
-            true,
-            true
-        )
-
-        val videoDecoderFactory = DefaultVideoDecoderFactory(mEGLBaseContext)
-
-
-        val audioDevice = createJavaAudioDevice()
-        val peerConnectionFactory = PeerConnectionFactory.builder()
-            .setAudioDeviceModule(audioDevice)
-            .setVideoEncoderFactory(videoEncoderFactory)
-            .setVideoDecoderFactory(videoDecoderFactory)
-            .createPeerConnectionFactory()
-        audioDevice.release()
-
-        return peerConnectionFactory
-
-    }
-
-    private fun createAudioConstraints(): MediaConstraints {
-        val audioConstraints = MediaConstraints()
-        audioConstraints.mandatory.add(
-            MediaConstraints.KeyValuePair(
-                "googEchoCancellation",
-                "true"
-            )
-        )
-        audioConstraints.mandatory.add(
-            MediaConstraints.KeyValuePair(
-                "googAutoGainControl",
-                "false"
-            )
-        )
-        audioConstraints.mandatory.add(
-            MediaConstraints.KeyValuePair(
-                "googHighpassFilter",
-                "true"
-            )
-        )
-        audioConstraints.mandatory.add(
-            MediaConstraints.KeyValuePair(
-                "googNoiseSuppression",
-                "true"
-            )
-        )
-        return audioConstraints
-    }
-
-    private fun offerOrAnswerConstraint(): MediaConstraints {
-        val mediaConstraints = MediaConstraints()
-        val keyValuePairs = java.util.ArrayList<MediaConstraints.KeyValuePair>()
-        keyValuePairs.add(MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"))
-        keyValuePairs.add(MediaConstraints.KeyValuePair("OfferToReceiveVideo", "true"))
-        mediaConstraints.mandatory.addAll(keyValuePairs)
-        return mediaConstraints
-    }
-
-    private fun createJavaAudioDevice(): AudioDeviceModule {
-        val audioTrackErrorCallback: JavaAudioDeviceModule.AudioTrackErrorCallback = object :
-            JavaAudioDeviceModule.AudioTrackErrorCallback {
-            override fun onWebRtcAudioTrackInitError(errorMessage: String) {
-                Log.i(TAG, "onWebRtcAudioTrackInitError ============> $errorMessage")
-
-            }
-
-            override fun onWebRtcAudioTrackStartError(
-                errorCode: JavaAudioDeviceModule.AudioTrackStartErrorCode, errorMessage: String
-            ) {
-                Log.i(TAG, "onWebRtcAudioTrackStartError ============> $errorCode:$errorMessage")
-
-            }
-
-            override fun onWebRtcAudioTrackError(errorMessage: String) {
-                Log.i(TAG, "onWebRtcAudioTrackError ============> $errorMessage")
-
-            }
-        }
-
-
-        // Set audio track state callbacks.
-        val audioTrackStateCallback: JavaAudioDeviceModule.AudioTrackStateCallback = object :
-            JavaAudioDeviceModule.AudioTrackStateCallback {
-            override fun onWebRtcAudioTrackStart() {
-                Log.i(TAG, "onWebRtcAudioTrackStart ============>")
-
-            }
-
-            override fun onWebRtcAudioTrackStop() {
-                Log.i(TAG, "onWebRtcAudioTrackStop ============>")
-
-            }
-        }
-
-        return JavaAudioDeviceModule.builder(mContext)
-            .setUseHardwareAcousticEchoCanceler(true)
-            .setUseHardwareNoiseSuppressor(true)
-            .setAudioTrackErrorCallback(audioTrackErrorCallback)
-            .setAudioTrackStateCallback(audioTrackStateCallback)
-            .setUseStereoOutput(true) //立体声
-            .createAudioDeviceModule()
-    }
-
-    fun setVideoPath(url: String) {
-        videoUrl = url
-    }
-
-    fun start() {
-
-        mLocalPeer = Peer {
-            val okHttpClient = OkHttpClient.Builder().build()
-
-
-            val body = RequestBody.create("text/plain; charset=utf-8".toMediaType(), it!!)
-
-
-            val request: Request = Request.Builder()
-                .url(videoUrl)
-                .post(body)
-                .build()
-
-            val call: Call = okHttpClient.newCall(request)
-
-            call.enqueue(object : Callback {
-                override fun onFailure(call: Call, e: IOException) {
-                    Log.i(TAG, "onFailure")
-                    OnErrorListener?.invoke(
-                        ErrorCode.GET_REMOTE_SDP_ERROR.errorCode,
-                        e.message.toString()
-                    )
-                }
-
-                override fun onResponse(call: Call, response: Response) {
-                    val body = response.body?.string()
-                    val sdpResponse = Gson().fromJson(body, SdpResponse::class.java)
-
-                    try {
-                        mRemoteSessionDescription = SessionDescription(
-                            SessionDescription.Type.fromCanonicalForm("answer"),
-                            sdpResponse.sdp
-                        )
-                        Log.i(
-                            TAG,
-                            "RemoteSdpObserver onCreateSuccess:[SessionDescription[type=${mRemoteSessionDescription?.type?.name},description=${mRemoteSessionDescription?.description}]]"
-                        )
-                        mLocalPeer?.setRemoteDescription(mRemoteSessionDescription!!)
-                    } catch (e: Exception) {
-                        Log.i(TAG, e.toString())
-                        OnErrorListener.invoke(
-                            ErrorCode.GET_REMOTE_SDP_ERROR.errorCode,
-                            e.localizedMessage
-                        )
-                    }
-                }
-            })
-        }
-    }
-
-    fun pause() {
-        mSurfaceViewRenderer.pauseVideo()
-        //mSurfaceViewRenderer.disableFpsReduction()
-    }
-
-    fun resume() {
-        mSurfaceViewRenderer.setFpsReduction(15f)
-    }
-
-    fun screenshot(listener: (bitmap: Bitmap) -> Unit) {
-        mSurfaceViewRenderer.addFrameListener({
-            listener.invoke(it)
-        }, 1f)
-    }
-
-    fun setSpeakerphoneOn(on: Boolean) {
-        audioManager.isSpeakerphoneOn = on
-    }
-
-    fun mute(on:Boolean) {
-        audioManager.isMicrophoneMute=on
-    }
-
-    override fun onDestroy(owner: LifecycleOwner) {
-        super.onDestroy(owner)
-        mSurfaceViewRenderer.release()
-        mLocalPeer?.mPeerConnection?.dispose()
-        mAudioSource?.dispose()
-        mPeerConnectionFactory?.dispose()
-    }
-
-    override fun onMeasure(widthMeasureSpec: Int, heightMeasureSpec: Int) {
-        super.onMeasure(widthMeasureSpec, heightMeasureSpec)
-
-    }
-
-    inner class Peer(var sdp: (String?) -> Unit = {}) : PeerConnection.Observer, SdpObserver {
-
-        var mPeerConnection: PeerConnection? = null
-
-        init {
-            mPeerConnection = createPeerConnection()
-            mPeerConnection?.createOffer(this, offerOrAnswerConstraint())
-        }
-
-        //初始化 RTCPeerConnection 连接管道
-        private fun createPeerConnection(): PeerConnection? {
-            if (mPeerConnectionFactory == null) {
-                mPeerConnectionFactory = createConnectionFactory()
-            }
-            // 管道连接抽象类实现方法
-            val ICEServers = LinkedList<PeerConnection.IceServer>()
-            val rtcConfig = PeerConnection.RTCConfiguration(ICEServers)
-            //修改模式 PlanB无法使用仅接收音视频的配置
-            //rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.PLAN_B
-            return mPeerConnectionFactory?.createPeerConnection(rtcConfig, this)
-        }
-
-        fun setRemoteDescription(sdp: SessionDescription) {
-            mPeerConnection?.setRemoteDescription(this, sdp)
-        }
-
-        override fun onCreateSuccess(sessionDescription: SessionDescription?) {
-            mPeerConnection?.setLocalDescription(this, sessionDescription)
-            mPeerConnection?.addStream(mLocalMediaStream)
-            sdp.invoke(sessionDescription?.description)
-        }
-
-        override fun onSetSuccess() {
-
-        }
-
-        override fun onCreateFailure(p0: String?) {
-
-        }
-
-        override fun onSetFailure(p0: String?) {
-
-        }
-
-        override fun onSignalingChange(signalingState: PeerConnection.SignalingState?) {
-            Log.i(TAG, "onSignalingChange ============> " + signalingState.toString())
-        }
-
-        override fun onIceConnectionChange(iceConnectionState: PeerConnection.IceConnectionState?) {
-            Log.i(TAG, "onIceConnectionChange ============> " + iceConnectionState.toString())
-
-        }
-
-        override fun onIceConnectionReceivingChange(p0: Boolean) {
-            Log.i(TAG, "onIceConnectionReceivingChange ============> $p0")
-
-        }
-
-        override fun onIceGatheringChange(iceGatheringState: PeerConnection.IceGatheringState?) {
-            Log.i(TAG, "onIceGatheringChange ============> ${iceGatheringState.toString()}")
-        }
-
-        override fun onIceCandidate(iceCandidate: IceCandidate?) {
-            Log.i(TAG, "onIceCandidate ============> ${iceCandidate.toString()}")
-
-
-        }
-
-        override fun onIceCandidatesRemoved(p0: Array<out IceCandidate>?) {
-            Log.i(TAG, "onIceCandidatesRemoved ============> ${p0.toString()}")
-        }
-
-        override fun onAddStream(mediaStream: MediaStream?) {
-            Log.i(TAG, "onAddStream ============> ${mediaStream?.toString()}")
-
-            if (mediaStream?.videoTracks?.isEmpty() != true) {
-                val remoteVideoTrack = mediaStream?.videoTracks?.get(0)
-                remoteVideoTrack?.setEnabled(true)
-                remoteVideoTrack?.addSink(mSurfaceViewRenderer)
-            }
-
-            if (mediaStream?.audioTracks?.isEmpty() != true) {
-                val remoteAudioTrack = mediaStream?.audioTracks?.get(0)
-                remoteAudioTrack?.setEnabled(true)
-                remoteAudioTrack?.setVolume(1.0)
-            }
-
-
-        }
-
-        override fun onRemoveStream(mediaStream: MediaStream?) {
-            Log.i(TAG, "onRemoveStream ============> ${mediaStream.toString()}")
-
-        }
-
-        override fun onDataChannel(dataChannel: DataChannel?) {
-            Log.i(TAG, "onDataChannel ============> ${dataChannel.toString()}")
-
-        }
-
-        override fun onRenegotiationNeeded() {
-            Log.i(TAG, "onRenegotiationNeeded ============>")
-
-        }
-
-        override fun onAddTrack(rtpReceiver: RtpReceiver?, p1: Array<out MediaStream>?) {
-            Log.i(TAG, "onAddTrack ============>" + rtpReceiver?.track())
-            Log.i(TAG, "onAddTrack ============>" + p1?.size)
-
-        }
-    }
-
-    override fun onFirstFrameRendered() {
-        Log.i(TAG, "onFirstFrameRendered ============>")
-
-    }
-
-    override fun onFrameResolutionChanged(frameWidth: Int, frameHeight: Int, rotation: Int) {
-        Log.i(TAG, "onFrameResolutionChanged ============> $frameWidth:$frameHeight:$rotation")
-        //set(frameWidth,frameHeight)
-    }
-
-
-
-
-}

webrtc_player/android/app/src/main/res/drawable-v24/ic_launcher_foreground.xml

@@ -1,30 +0,0 @@
-<vector xmlns:android="http://schemas.android.com/apk/res/android"
-    xmlns:aapt="http://schemas.android.com/aapt"
-    android:width="108dp"
-    android:height="108dp"
-    android:viewportWidth="108"
-    android:viewportHeight="108">
-    <path android:pathData="M31,63.928c0,0 6.4,-11 12.1,-13.1c7.2,-2.6 26,-1.4 26,-1.4l38.1,38.1L107,108.928l-32,-1L31,63.928z">
-        <aapt:attr name="android:fillColor">
-            <gradient
-                android:endX="85.84757"
-                android:endY="92.4963"
-                android:startX="42.9492"
-                android:startY="49.59793"
-                android:type="linear">
-                <item
-                    android:color="#44000000"
-                    android:offset="0.0" />
-                <item
-                    android:color="#00000000"
-                    android:offset="1.0" />
-            </gradient>
-        </aapt:attr>
-    </path>
-    <path
-        android:fillColor="#FFFFFF"
-        android:fillType="nonZero"
-        android:pathData="M65.3,45.828l3.8,-6.6c0.2,-0.4 0.1,-0.9 -0.3,-1.1c-0.4,-0.2 -0.9,-0.1 -1.1,0.3l-3.9,6.7c-6.3,-2.8 -13.4,-2.8 -19.7,0l-3.9,-6.7c-0.2,-0.4 -0.7,-0.5 -1.1,-0.3C38.8,38.328 38.7,38.828 38.9,39.228l3.8,6.6C36.2,49.428 31.7,56.028 31,63.928h46C76.3,56.028 71.8,49.428 65.3,45.828zM43.4,57.328c-0.8,0 -1.5,-0.5 -1.8,-1.2c-0.3,-0.7 -0.1,-1.5 0.4,-2.1c0.5,-0.5 1.4,-0.7 2.1,-0.4c0.7,0.3 1.2,1 1.2,1.8C45.3,56.528 44.5,57.328 43.4,57.328L43.4,57.328zM64.6,57.328c-0.8,0 -1.5,-0.5 -1.8,-1.2s-0.1,-1.5 0.4,-2.1c0.5,-0.5 1.4,-0.7 2.1,-0.4c0.7,0.3 1.2,1 1.2,1.8C66.5,56.528 65.6,57.328 64.6,57.328L64.6,57.328z"
-        android:strokeWidth="1"
-        android:strokeColor="#00000000" />
-</vector>

webrtc_player/android/app/src/main/res/drawable/ic_launcher_background.xml

@@ -1,170 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<vector xmlns:android="http://schemas.android.com/apk/res/android"
-    android:width="108dp"
-    android:height="108dp"
-    android:viewportWidth="108"
-    android:viewportHeight="108">
-    <path
-        android:fillColor="#3DDC84"
-        android:pathData="M0,0h108v108h-108z" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M9,0L9,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M19,0L19,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M29,0L29,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M39,0L39,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M49,0L49,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M59,0L59,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M69,0L69,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M79,0L79,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M89,0L89,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M99,0L99,108"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,9L108,9"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,19L108,19"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,29L108,29"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,39L108,39"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,49L108,49"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,59L108,59"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,69L108,69"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,79L108,79"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,89L108,89"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M0,99L108,99"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M19,29L89,29"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M19,39L89,39"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M19,49L89,49"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M19,59L89,59"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M19,69L89,69"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M19,79L89,79"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M29,19L29,89"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M39,19L39,89"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M49,19L49,89"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M59,19L59,89"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M69,19L69,89"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-    <path
-        android:fillColor="#00000000"
-        android:pathData="M79,19L79,89"
-        android:strokeWidth="0.8"
-        android:strokeColor="#33FFFFFF" />
-</vector>

webrtc_player/android/app/src/main/res/layout/activity_main.xml

@@ -1,93 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<androidx.constraintlayout.widget.ConstraintLayout xmlns:android="http://schemas.android.com/apk/res/android"
-    xmlns:app="http://schemas.android.com/apk/res-auto"
-    xmlns:tools="http://schemas.android.com/tools"
-    android:layout_width="match_parent"
-    android:layout_height="match_parent"
-    tools:context=".MainActivity">
-
-    <com.zlmediakit.webrtc.WebRTCSurfaceView
-        android:id="@+id/web_rtc_sv"
-        android:layout_width="match_parent"
-        android:layout_height="200dp"
-        app:layout_constraintTop_toTopOf="parent" />
-
-
-    <androidx.appcompat.widget.AppCompatEditText
-        android:id="@+id/url"
-        android:layout_width="match_parent"
-        android:layout_height="wrap_content"
-        app:layout_constraintTop_toBottomOf="@+id/web_rtc_sv"
-        android:text=""/>
-    <LinearLayout
-        android:id="@+id/ll"
-        android:layout_width="match_parent"
-        android:layout_height="wrap_content"
-        android:orientation="horizontal"
-        android:layout_marginTop="30dp"
-        app:layout_constraintTop_toBottomOf="@+id/url">
-
-        <androidx.appcompat.widget.AppCompatButton
-            android:id="@+id/btn_play"
-            android:layout_width="wrap_content"
-            android:layout_height="wrap_content"
-            android:text="播放" />
-        <androidx.appcompat.widget.AppCompatButton
-            android:id="@+id/btn_pause"
-            android:layout_width="wrap_content"
-            android:layout_height="wrap_content"
-            android:text="暂停" />
-
-        <androidx.appcompat.widget.AppCompatButton
-            android:id="@+id/btn_resume"
-            android:layout_width="wrap_content"
-            android:layout_height="wrap_content"
-            android:text="恢复" />
-
-        <androidx.appcompat.widget.AppCompatButton
-            android:id="@+id/btn_speaker"
-            android:layout_width="wrap_content"
-            android:layout_height="wrap_content"
-            android:text="扬声器" />
-
-        <androidx.appcompat.widget.AppCompatButton
-            android:id="@+id/btn_mute"
-            android:layout_width="wrap_content"
-            android:layout_height="wrap_content"
-            android:text="静音" />
-
-    </LinearLayout>
-
-    <LinearLayout
-        android:id="@+id/ll2"
-        android:layout_width="match_parent"
-        android:layout_height="wrap_content"
-        android:orientation="horizontal"
-        android:layout_marginTop="10dp"
-        app:layout_constraintTop_toBottomOf="@+id/ll">
-
-
-        <androidx.appcompat.widget.AppCompatButton
-            android:id="@+id/btn_screenshot"
-            android:layout_width="wrap_content"
-            android:layout_height="wrap_content"
-            android:text="截图" />
-
-        <androidx.appcompat.widget.AppCompatButton
-            android:id="@+id/btn_screen_record"
-            android:layout_width="wrap_content"
-            android:layout_height="wrap_content"
-            android:text="录制" />
-
-
-    </LinearLayout>
-
-
-    <androidx.appcompat.widget.AppCompatImageView
-        android:id="@+id/iv_screen"
-        android:layout_width="wrap_content"
-        android:layout_height="wrap_content"
-        app:layout_constraintBottom_toBottomOf="parent"
-        tools:ignore="MissingConstraints" />
-
-</androidx.constraintlayout.widget.ConstraintLayout>

webrtc_player/android/app/src/main/res/layout/layout_videoview.xml

@@ -1,13 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android"
-    xmlns:tools="http://schemas.android.com/tools"
-    android:layout_width="match_parent"
-    android:layout_height="match_parent"
-    xmlns:app="http://schemas.android.com/apk/res-auto">
-
-    <org.webrtc.SurfaceViewRenderer
-        android:id="@+id/surface_view_renderer"
-        android:layout_width="wrap_content"
-        android:layout_height="wrap_content"/>
-
-</RelativeLayout>

webrtc_player/android/app/src/main/res/mipmap-anydpi-v26/ic_launcher.xml

@@ -1,5 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<adaptive-icon xmlns:android="http://schemas.android.com/apk/res/android">
-    <background android:drawable="@drawable/ic_launcher_background" />
-    <foreground android:drawable="@drawable/ic_launcher_foreground" />
-</adaptive-icon>

webrtc_player/android/app/src/main/res/mipmap-anydpi-v26/ic_launcher_round.xml

@@ -1,5 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<adaptive-icon xmlns:android="http://schemas.android.com/apk/res/android">
-    <background android:drawable="@drawable/ic_launcher_background" />
-    <foreground android:drawable="@drawable/ic_launcher_foreground" />
-</adaptive-icon>

webrtc_player/android/app/src/main/res/values-night/themes.xml

@@ -1,16 +0,0 @@
-<resources xmlns:tools="http://schemas.android.com/tools">
-    <!-- Base application theme. -->
-    <style name="Theme.AndroidWebRTC" parent="Theme.MaterialComponents.DayNight.DarkActionBar">
-        <!-- Primary brand color. -->
-        <item name="colorPrimary">@color/purple_200</item>
-        <item name="colorPrimaryVariant">@color/purple_700</item>
-        <item name="colorOnPrimary">@color/black</item>
-        <!-- Secondary brand color. -->
-        <item name="colorSecondary">@color/teal_200</item>
-        <item name="colorSecondaryVariant">@color/teal_200</item>
-        <item name="colorOnSecondary">@color/black</item>
-        <!-- Status bar color. -->
-        <item name="android:statusBarColor" tools:targetApi="l">?attr/colorPrimaryVariant</item>
-        <!-- Customize your theme here. -->
-    </style>
-</resources>

webrtc_player/android/app/src/main/res/values/colors.xml

@@ -1,10 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<resources>
-    <color name="purple_200">#FFBB86FC</color>
-    <color name="purple_500">#FF6200EE</color>
-    <color name="purple_700">#FF3700B3</color>
-    <color name="teal_200">#FF03DAC5</color>
-    <color name="teal_700">#FF018786</color>
-    <color name="black">#FF000000</color>
-    <color name="white">#FFFFFFFF</color>
-</resources>

webrtc_player/android/app/src/main/res/values/strings.xml

@@ -1,3 +0,0 @@
-<resources>
-    <string name="app_name">AndroidWebRTC</string>
-</resources>

webrtc_player/android/app/src/main/res/values/themes.xml

@@ -1,16 +0,0 @@
-<resources xmlns:tools="http://schemas.android.com/tools">
-    <!-- Base application theme. -->
-    <style name="Theme.AndroidWebRTC" parent="Theme.MaterialComponents.DayNight.DarkActionBar">
-        <!-- Primary brand color. -->
-        <item name="colorPrimary">@color/purple_500</item>
-        <item name="colorPrimaryVariant">@color/purple_700</item>
-        <item name="colorOnPrimary">@color/white</item>
-        <!-- Secondary brand color. -->
-        <item name="colorSecondary">@color/teal_200</item>
-        <item name="colorSecondaryVariant">@color/teal_700</item>
-        <item name="colorOnSecondary">@color/black</item>
-        <!-- Status bar color. -->
-        <item name="android:statusBarColor" tools:targetApi="l">?attr/colorPrimaryVariant</item>
-        <!-- Customize your theme here. -->
-    </style>
-</resources>

webrtc_player/android/app/src/main/res/xml/backup_rules.xml

@@ -1,13 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?><!--
-   Sample backup rules file; uncomment and customize as necessary.
-   See https://developer.android.com/guide/topics/data/autobackup
-   for details.
-   Note: This file is ignored for devices older that API 31
-   See https://developer.android.com/about/versions/12/backup-restore
--->
-<full-backup-content>
-    <!--
-   <include domain="sharedpref" path="."/>
-   <exclude domain="sharedpref" path="device.xml"/>
--->
-</full-backup-content>

webrtc_player/android/app/src/main/res/xml/data_extraction_rules.xml

@@ -1,19 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?><!--
-   Sample data extraction rules file; uncomment and customize as necessary.
-   See https://developer.android.com/about/versions/12/backup-restore#xml-changes
-   for details.
--->
-<data-extraction-rules>
-    <cloud-backup>
-        <!-- TODO: Use <include> and <exclude> to control what is backed up.
-        <include .../>
-        <exclude .../>
-        -->
-    </cloud-backup>
-    <!--
-    <device-transfer>
-        <include .../>
-        <exclude .../>
-    </device-transfer>
-    -->
-</data-extraction-rules>

webrtc_player/android/app/src/test/java/com/zlmediakit/webrtc/ExampleUnitTest.kt

@@ -1,17 +0,0 @@
-package com.zlmediakit.webrtc
-
-import org.junit.Test
-
-import org.junit.Assert.*
-
-/**
- * Example local unit test, which will execute on the development machine (host).
- *
- * See [testing documentation](http://d.android.com/tools/testing).
- */
-class ExampleUnitTest {
-    @Test
-    fun addition_isCorrect() {
-        assertEquals(4, 2 + 2)
-    }
-}

webrtc_player/android/build.gradle

@@ -1,19 +0,0 @@
-// Top-level build file where you can add configuration options common to all sub-projects/modules.
-buildscript {
-    ext.kotlin_version = '1.7.10'
-    repositories {
-        mavenCentral()
-    }
-    dependencies {
-        classpath "org.jetbrains.kotlin:kotlin-gradle-plugin:$kotlin_version"
-    }
-}
-plugins {
-    id 'com.android.application' version '7.2.1' apply false
-    id 'com.android.library' version '7.2.1' apply false
-    id 'org.jetbrains.kotlin.android' version '1.7.10' apply false
-}
-
-task clean(type: Delete) {
-    delete rootProject.buildDir
-}

webrtc_player/android/gradle.properties

@@ -1,23 +0,0 @@
-# Project-wide Gradle settings.
-# IDE (e.g. Android Studio) users:
-# Gradle settings configured through the IDE *will override*
-# any settings specified in this file.
-# For more details on how to configure your build environment visit
-# http://www.gradle.org/docs/current/userguide/build_environment.html
-# Specifies the JVM arguments used for the daemon process.
-# The setting is particularly useful for tweaking memory settings.
-org.gradle.jvmargs=-Xmx2048m -Dfile.encoding=UTF-8
-# When configured, Gradle will run in incubating parallel mode.
-# This option should only be used with decoupled projects. More details, visit
-# http://www.gradle.org/docs/current/userguide/multi_project_builds.html#sec:decoupled_projects
-# org.gradle.parallel=true
-# AndroidX package structure to make it clearer which packages are bundled with the
-# Android operating system, and which are packaged with your app"s APK
-# https://developer.android.com/topic/libraries/support-library/androidx-rn
-android.useAndroidX=true
-# Kotlin code style for this project: "official" or "obsolete":
-kotlin.code.style=official
-# Enables namespacing of each library's R class so that its R class includes only the
-# resources declared in the library itself and none from the library's dependencies,
-# thereby reducing the size of the R class for that library
-android.nonTransitiveRClass=true