android rtc播放、推流基本ok

webrtc_player/android/.gitattributes

@@ -1,2 +0,0 @@
-zlm/libs/arm64-v8a/libZLToolKit.a filter=lfs diff=lfs merge=lfs -text
-zlm/libs/armeabi-v7a/libZLToolKit.a filter=lfs diff=lfs merge=lfs -text

webrtc_player/android/.gitignore

@@ -13,4 +13,3 @@
 .externalNativeBuild
 .cxx
 local.properties
-/.idea/

webrtc_player/android/.idea/.gitignore

@@ -0,0 +1,3 @@
+# Default ignored files
+/shelf/
+/workspace.xml

webrtc_player/android/.idea/.name

@@ -0,0 +1 @@
+RTCPlayer

webrtc_player/android/.idea/compiler.xml

@@ -0,0 +1,6 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project version="4">
+  <component name="CompilerConfiguration">
+    <bytecodeTargetLevel target="17" />
+  </component>
+</project>

webrtc_player/android/.idea/deploymentTargetDropDown.xml

@@ -0,0 +1,10 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project version="4">
+  <component name="deploymentTargetDropDown">
+    <value>
+      <entry key="app">
+        <State />
+      </entry>
+    </value>
+  </component>
+</project>

webrtc_player/android/.idea/gradle.xml

@@ -0,0 +1,18 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project version="4">
+  <component name="GradleSettings">
+    <option name="linkedExternalProjectsSettings">
+      <GradleProjectSettings>
+        <option name="externalProjectPath" value="$PROJECT_DIR$" />
+        <option name="gradleJvm" value="#GRADLE_LOCAL_JAVA_HOME" />
+        <option name="modules">
+          <set>
+            <option value="$PROJECT_DIR$" />
+            <option value="$PROJECT_DIR$/app" />
+          </set>
+        </option>
+        <option name="resolveExternalAnnotations" value="false" />
+      </GradleProjectSettings>
+    </option>
+  </component>
+</project>

webrtc_player/android/.idea/kotlinc.xml

@@ -0,0 +1,6 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project version="4">
+  <component name="KotlinJpsPluginSettings">
+    <option name="version" value="1.9.22" />
+  </component>
+</project>

webrtc_player/android/.idea/migrations.xml

@@ -0,0 +1,10 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project version="4">
+  <component name="ProjectMigrations">
+    <option name="MigrateToGradleLocalJavaHome">
+      <set>
+        <option value="$PROJECT_DIR$" />
+      </set>
+    </option>
+  </component>
+</project>

webrtc_player/android/.idea/misc.xml

@@ -0,0 +1,10 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project version="4">
+  <component name="ExternalStorageConfigurationManager" enabled="true" />
+  <component name="ProjectRootManager" version="2" languageLevel="JDK_17" default="true" project-jdk-name="jbr-17" project-jdk-type="JavaSDK">
+    <output url="file://$PROJECT_DIR$/build/classes" />
+  </component>
+  <component name="ProjectType">
+    <option name="id" value="Android" />
+  </component>
+</project>

webrtc_player/android/.idea/vcs.xml

@@ -0,0 +1,6 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project version="4">
+  <component name="VcsDirectoryMappings">
+    <mapping directory="$PROJECT_DIR$/../.." vcs="Git" />
+  </component>
+</project>

webrtc_player/android/README.md

@@ -0,0 +1,104 @@
+# RTCPlayer 播放器
+
+一个基于Android客户端的的RTC播放器
+
+## 项目特点
+
+  - 低延迟
+  - [X]支持[ZLMediakit](https://github.com/ZLMediaKit/ZLMediaKit)流媒体
+  - [ ]支持[SRS](https://github.com/ossrs/srs)
+  - [ ]支持[Janus](https://github.com/meetecho/janus-gateway)
+
+<center class="half">
+    <img src="doc/home.jpg" width="240"/>&nbsp;&nbsp;&nbsp;
+    <img src="doc/player.jpg" width="240"/>&nbsp;&nbsp;&nbsp;
+    <img src="doc/pusher.jpg" width="240"/>
+</center>
+
+## 延迟情况
+
+- **网页端推流，Android端播放**
+
+<center class="half">
+    <img src="doc/delay_play1.jpg" width="240"/>&nbsp;&nbsp;&nbsp;
+    <img src="doc/delay_play2.jpg" width="240"/>&nbsp;&nbsp;&nbsp;
+    <img src="doc/delay_play3.jpg" width="240"/>
+</center>
+
+三次延迟分别为：490ms、526ms、560ms
+
+- **Android端推屏幕流，网页端播放**
+
+<center class="half">
+    <img src="doc/delay_push1.jpg" width="240"/>&nbsp;&nbsp;&nbsp;
+    <img src="doc/delay_push2.jpg" width="240"/>&nbsp;&nbsp;&nbsp;
+    <img src="doc/delay_push3.jpg" width="240"/>
+</center>
+
+三次延迟分别为：440ms、430ms、387ms
+
+## 接口说明
+
+### Player
+
+- bind(surface: RTCSurfaceView)
+
+  绑定视图
+
+- play(app: String, streamId: String)
+
+  播放
+- setVolume(volume:Float)
+
+  设置音量，范围: 0.0~1.0f
+- stop()
+
+  停止播放
+- pause()
+
+  暂停播放
+- resume()
+
+  恢复播放
+- capture(listener: (bitmap: Bitmap) -> Unit)
+
+  截图
+- record(duration: Long, result: (path: String) -> Unit)
+
+  录制
+- setOnErrorListener(listener: (code: Int, msg: String) -> Unit)
+  
+  设置播放器错误监听回调
+- setOnStatusListener(listener: (status: Status) -> Unit)
+
+  设置播放器状态回调
+
+### Pusher
+
+- bind(surface: RTCSurfaceView, localPreview: Boolean)
+
+  绑定视图
+
+- push(app: String, streamId: String, mode: PushMode = PushMode.CAMERA, inputFile: String = "")
+
+  推流，支持Camera、Screen、File
+
+- stop()
+  
+  停止推流
+
+- setOnErrorListener(listener: (code: Int, msg: String) -> Unit)
+
+  设置播放器错误监听回调
+
+## 联系作者
+如果您需要深度二次开发，并支持其他流媒体服务，可以找我哦！
+[李之阳](https://github.com/leo94666)
+
+## 特别感谢
+
+感谢[ZLMediakit](https://github.com/ZLMediaKit/ZLMediaKit)开源项目
+
+同时感谢JetBrains对开源项目的支持，本项目使用Android Studio开发与调试：
+
+[![Android Studio](https://th.bing.com/th?id=ODLS.d2ea10a5-5792-4f82-bd13-1595fd9d969c&w=32&h=32&qlt=90&pcl=fffffa&o=6&pid=1.2)](https://developer.android.com/studio?hl=zh-cn)

webrtc_player/android/app/.gitignore

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

webrtc_player/android/app/build.gradle

@@ -1,56 +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 24
-        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 project(':zlm')
-    implementation 'com.guolindev.permissionx:permissionx:1.7.1'
-
-
-}

webrtc_player/android/app/build.gradle.kts

@@ -0,0 +1,58 @@
+plugins {
+    id("com.android.application")
+    id("org.jetbrains.kotlin.android")
+
+}
+
+android {
+    namespace = "com.top.player"
+    compileSdk = 34
+
+    defaultConfig {
+        applicationId = "com.top.player"
+        minSdk = 24
+        targetSdk = 34
+        versionCode = 1
+        versionName = "1.0"
+
+        testInstrumentationRunner = "androidx.test.runner.AndroidJUnitRunner"
+
+    }
+
+    buildTypes {
+        release {
+            isMinifyEnabled = 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"
+    }
+    viewBinding {
+        enable = true
+    }
+}
+
+dependencies {
+
+    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("org.jetbrains.kotlin:kotlin-stdlib-jdk7:1.8.22")
+    implementation("com.guolindev.permissionx:permissionx:1.7.1")
+
+    //implementation(project(":RTCPlayer"))
+    implementation("com.rtc.core:RTCPlayer:1.0.0.beta")
+
+}

webrtc_player/android/app/src/androidTest/java/com/top/player/ExampleInstrumentedTest.java

@@ -0,0 +1,26 @@
+package com.top.player;
+
+import android.content.Context;
+
+import androidx.test.platform.app.InstrumentationRegistry;
+import androidx.test.ext.junit.runners.AndroidJUnit4;
+
+import org.junit.Test;
+import org.junit.runner.RunWith;
+
+import static org.junit.Assert.*;
+
+/**
+ * Instrumented test, which will execute on an Android device.
+ *
+ * @see <a href="http://d.android.com/tools/testing">Testing documentation</a>
+ */
+@RunWith(AndroidJUnit4.class)
+public class ExampleInstrumentedTest {
+    @Test
+    public void useAppContext() {
+        // Context of the app under test.
+        Context appContext = InstrumentationRegistry.getInstrumentation().getTargetContext();
+        assertEquals("com.top.player", appContext.getPackageName());
+    }
+}

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

@@ -1,24 +1,27 @@
 <?xml version="1.0" encoding="utf-8"?>
 <manifest xmlns:android="http://schemas.android.com/apk/res/android"
-    xmlns:tools="http://schemas.android.com/tools"
+    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" />
-    <uses-feature android:name="android.hardware.camera.autofocus"/>
+    <uses-feature android:name="android.hardware.camera.autofocus" />
     <uses-feature
         android:glEsVersion="0x00020000"
-        android:required="true"/>
+        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" />
+
+    <uses-permission android:name="android.permission.FOREGROUND_SERVICE"/>
+    <uses-permission android:name="android.permission.FOREGROUND_SERVICE_MEDIA_PROJECTION" />
 
-    <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"
@@ -27,15 +30,14 @@
         android:label="@string/app_name"
         android:roundIcon="@mipmap/ic_launcher_round"
         android:supportsRtl="true"
-        android:theme="@style/Theme.AndroidWebRTC"
+        android:theme="@style/Theme.RTCPlayer"
-        tools:overrideLibrary="com.zlm.rtc"
+        tools:targetApi="31"
         android:usesCleartextTraffic="true"
         android:hardwareAccelerated="true"
-        android:name=".App"
+        tools:overrideLibrary="com.rtc.core"
-        tools:targetApi="31">
+        android:name=".App">
         <activity
             android:name=".MainActivity"
-            android:screenOrientation="portrait"
             android:exported="true">
             <intent-filter>
                 <action android:name="android.intent.action.MAIN" />
@@ -43,12 +45,13 @@
                 <category android:name="android.intent.category.LAUNCHER" />
             </intent-filter>
         </activity>
-        <activity android:name=".PlayDemoActivity"
+
-            android:screenOrientation="portrait"/>
+        <activity
-        <activity android:name=".PlayerDemoActivity"
+            android:name=".PlayerDemoActivity"
-            android:screenOrientation="portrait"/>
+            android:screenOrientation="portrait" />
-        <activity android:name=".PusherDemoActivity"
+        <activity
-            android:screenOrientation="portrait"/>
+            android:name=".PusherDemoActivity"
+            android:screenOrientation="portrait" />
     </application>
 
 </manifest>

webrtc_player/android/app/src/main/java/com/top/player/App.kt

@@ -1,4 +1,4 @@
-package com.zlmediakit.webrtc
+package com.top.player
 
 import android.app.Application
 

webrtc_player/android/app/src/main/java/com/top/player/MainActivity.kt

@@ -1,4 +1,4 @@
-package com.zlmediakit.webrtc
+package com.top.player
 
 import android.content.Intent
 import android.os.Bundle
@@ -11,7 +11,6 @@ class MainActivity : AppCompatActivity() {
         super.onCreate(savedInstanceState)
         setContentView(R.layout.activity_main)
 
-
     }
 
     fun toPlayActivity(view: View) {
@@ -23,7 +22,4 @@ class MainActivity : AppCompatActivity() {
 
     }
 
-    fun toDataChannelActivity(view: View) {
-
-    }
 }

webrtc_player/android/app/src/main/java/com/top/player/PlayerDemoActivity.kt

@@ -0,0 +1,112 @@
+package com.top.player
+
+import android.os.Bundle
+import android.view.View
+import android.widget.Toast
+import androidx.appcompat.app.AppCompatActivity
+import com.rtc.core.ZLMRTCPlayer
+import com.rtc.core.play.Status
+import com.rtc.core.play.ZLMRTCPlayerImpl
+import com.top.player.databinding.ActivityPlayerBinding
+
+
+class PlayerDemoActivity : AppCompatActivity() {
+
+
+    private val player: ZLMRTCPlayer by lazy {
+        ZLMRTCPlayerImpl(this)
+    }
+
+
+    private val binding by lazy {
+        ActivityPlayerBinding.inflate(layoutInflater)
+    }
+
+    override fun onCreate(savedInstanceState: Bundle?) {
+        super.onCreate(savedInstanceState)
+
+        setContentView(binding.root)
+
+        //ffmpeg -re -stream_loop -1 -i "D:\li\hot\data\data\baseline.mp4" -vcodec h264 -acodec aac -f rtsp -rtsp_transport tcp -bf 0 rtsp://zlmediakit.com/live/li
+        //ffmpeg -re -stream_loop -1 -i "D:\li\hot\data\data\test.mp4" -vcodec h264 -acodec aac -f flv -bf 0 rtmp://zlmediakit.com/live/li
+
+        setTitle("Player Demo")
+        player.bind(binding.surfaceViewRender)
+
+        player.setOnErrorListener { code, msg ->
+            Toast.makeText(this, "code:$code,msg:${msg}", Toast.LENGTH_SHORT).show()
+
+        }
+
+        player.setOnStatusListener {
+            when (it) {
+                Status.PREPARING -> {
+                    binding.tvStatus.text = "准备播放"
+                }
+
+                Status.PLAYING -> {
+                    binding.tvStatus.text = "播放中.."
+                }
+
+                Status.PAUSE -> {
+                    binding.tvStatus.text = "暂停中.."
+                }
+
+                Status.RESUME -> {
+                    binding.tvStatus.text = "播放中.."
+                }
+
+                Status.STOP -> {
+                    binding.tvStatus.text = ""
+                }
+
+                Status.ERROR -> {
+                    binding.tvStatus.text = "播放异常"
+                }
+
+                else -> {}
+            }
+        }
+
+    }
+
+
+    override fun onDestroy() {
+        super.onDestroy()
+        player.stop()
+    }
+
+    fun onPlayClick(view: View) {
+
+        player.play(binding.tvApp.text.toString(), binding.tvStreamId.text.toString())
+    }
+
+    fun onPauseClick(view: View) {
+        player.pause()
+        //Toast.makeText(this, "ok", Toast.LENGTH_SHORT).show()
+    }
+
+    fun onStopClick(view: View) {
+        player.stop()
+    }
+
+    fun onResumeClick(view: View) {
+        player.resume()
+    }
+
+    fun onCapture(view: View) {
+        player.capture {
+            Toast.makeText(this, "capture ok", Toast.LENGTH_SHORT).show()
+        }
+    }
+
+    fun onRecord(view: View) {
+        player.record(10 * 1000) {
+            Toast.makeText(this, "" + it, Toast.LENGTH_SHORT).show()
+        }
+    }
+
+    fun onVolume(view: View) {
+        player.setVolume(0.0f)
+    }
+}

webrtc_player/android/app/src/main/java/com/top/player/PusherDemoActivity.kt

@@ -1,21 +1,19 @@
-package com.zlmediakit.webrtc
+package com.top.player
 
 import android.Manifest
 import android.content.Intent
-import android.content.pm.PackageManager
 import android.os.Bundle
 import android.view.View
+import android.widget.Toast
 import androidx.appcompat.app.AppCompatActivity
-import androidx.core.app.ActivityCompat
-import androidx.core.content.ContextCompat
 import com.permissionx.guolindev.PermissionX
-import com.permissionx.guolindev.callback.RequestCallback
+import com.rtc.core.RTCSurfaceView
-import com.zlm.rtc.ZLMRTCPusher
+import com.rtc.core.ZLMRTCPusher
-import com.zlm.rtc.push.PushMode
+import com.rtc.core.push.PushMode
-import com.zlm.rtc.push.ZLMRTCPusherImpl
+import com.rtc.core.push.ZLMRTCPusherImpl
-import kotlinx.android.synthetic.main.activity_player.surface_view_renderer
+import com.top.player.databinding.ActivityPlayerBinding
-import kotlinx.android.synthetic.main.activity_player.tv_app
+import com.top.player.databinding.ActivityPusherBinding
-import kotlinx.android.synthetic.main.activity_player.tv_stream_id
+
 
 class PusherDemoActivity : AppCompatActivity() {
 
@@ -25,15 +23,25 @@ class PusherDemoActivity : AppCompatActivity() {
     }
 
 
-    override fun onActivityResult(requestCode: Int, resultCode: Int, data: Intent?) {
+    private val binding by lazy {
-        super.onActivityResult(requestCode, resultCode, data)
+        ActivityPusherBinding.inflate(layoutInflater)
     }
+
+
     override fun onCreate(savedInstanceState: Bundle?) {
         super.onCreate(savedInstanceState)
 
         setContentView(R.layout.activity_pusher)
 
-        pusher.bind(surface_view_renderer, true)
+        val rtcSurfaceView = findViewById<RTCSurfaceView>(R.id.surface_view_render)
+
+        pusher.bind(rtcSurfaceView, true)
+        setTitle("Pusher Demo")
+
+
+        pusher.setOnErrorListener { code, msg ->
+            Toast.makeText(this, "code:${code},msg:${msg}",    Toast.LENGTH_SHORT).show()
+        }
 
     }
 
@@ -42,7 +50,7 @@ class PusherDemoActivity : AppCompatActivity() {
             .permissions(Manifest.permission.CAMERA, Manifest.permission.RECORD_AUDIO)
             .request { allGranted, grantedList, deniedList ->
                 if (allGranted) {
-                    pusher.push(tv_app.text.toString(), tv_stream_id.text.toString())
+                    pusher.push(binding.tvApp.text.toString(), binding.tvStreamId.text.toString())
                 }
             }
     }
@@ -53,8 +61,8 @@ class PusherDemoActivity : AppCompatActivity() {
             .request { allGranted, grantedList, deniedList ->
                 if (allGranted) {
                     pusher.push(
-                        tv_app.text.toString(),
+                        binding.tvApp.text.toString(),
-                        tv_stream_id.text.toString(),
+                        binding.tvStreamId.text.toString(),
                         PushMode.SCREEN
                     )
                 }
@@ -67,8 +75,8 @@ class PusherDemoActivity : AppCompatActivity() {
             .request { allGranted, grantedList, deniedList ->
                 if (allGranted) {
                     pusher.push(
-                        tv_app.text.toString(),
+                        binding.tvApp.text.toString(),
-                        tv_stream_id.text.toString(),
+                        binding.tvStreamId.text.toString(),
                         PushMode.FILE,
                         ""
                     )
@@ -81,5 +89,9 @@ class PusherDemoActivity : AppCompatActivity() {
         pusher.stop()
     }
 
+    fun onStopPush(view: View) {
+        pusher.stop()
+    }
+
 
 }

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

@@ -1,84 +0,0 @@
-package com.zlmediakit.webrtc
-
-import android.annotation.SuppressLint
-import android.graphics.drawable.BitmapDrawable
-import android.os.Bundle
-import android.widget.Toast
-import androidx.appcompat.app.AppCompatActivity
-import com.zlm.rtc.ZLMRTCPlayer
-import kotlinx.android.synthetic.main.activity_main.*
-import kotlinx.android.synthetic.main.activity_main.view.*
-import kotlinx.android.synthetic.main.activity_play.*
-
-
-class PlayDemoActivity : AppCompatActivity() {
-
-    private var isSpeaker = true
-
-    @SuppressLint("SetTextI18n")
-    override fun onCreate(savedInstanceState: Bundle?) {
-        super.onCreate(savedInstanceState)
-        setContentView(R.layout.activity_play)
-
-        lifecycle.addObserver(web_rtc_sv)
-
-
-
-
-
-        //http://124.223.98.45/index/api/webrtc?app=live&stream=test&type=play
-        url.setText("https://zlmediakit.com/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/PlayerDemoActivity.kt

@@ -1,70 +0,0 @@
-package com.zlmediakit.webrtc
-
-import android.os.Bundle
-import android.view.View
-import android.widget.Toast
-import androidx.appcompat.app.AppCompatActivity
-import com.zlm.rtc.ZLMRTCPlayer
-import com.zlm.rtc.play.ZLMRTCPlayerImpl
-import kotlinx.android.synthetic.main.activity_player.surface_view_renderer
-import kotlinx.android.synthetic.main.activity_player.tv_app
-import kotlinx.android.synthetic.main.activity_player.tv_stream_id
-
-class PlayerDemoActivity : AppCompatActivity() {
-
-
-    private val player: ZLMRTCPlayer by lazy {
-        ZLMRTCPlayerImpl(this)
-    }
-
-    override fun onCreate(savedInstanceState: Bundle?) {
-        super.onCreate(savedInstanceState)
-
-        setContentView(R.layout.activity_player)
-
-        //ffmpeg -re -stream_loop -1 -i "D:\li\hot\data\data\baseline.mp4" -vcodec h264 -acodec aac -f rtsp -rtsp_transport tcp -bf 0 rtsp://zlmediakit.com/live/li
-        //ffmpeg -re -stream_loop -1 -i "D:\li\hot\data\data\test.mp4" -vcodec h264 -acodec aac -f flv -bf 0 rtmp://zlmediakit.com/live/li
-
-        player.bind(surface_view_renderer)
-
-    }
-
-
-    override fun onDestroy() {
-        super.onDestroy()
-        player.stop()
-    }
-
-    fun onPlayClick(view: View) {
-
-        player.play(tv_app.text.toString(), tv_stream_id.text.toString())
-    }
-
-    fun onPauseClick(view: View) {
-        player.pause()
-    }
-
-    fun onStopClick(view: View) {
-        player.stop()
-    }
-
-    fun onResumeClick(view: View) {
-        player.resume()
-    }
-
-    fun onCapture(view: View) {
-        player.capture {
-            Toast.makeText(this, "capture ok", Toast.LENGTH_SHORT).show()
-        }
-    }
-
-    fun onRecord(view: View) {
-        player.record(10 * 1000) {
-            Toast.makeText(this, "" + it, Toast.LENGTH_SHORT).show()
-        }
-    }
-
-    fun onVolume(view: View) {
-        player.setVolume()
-    }
-}

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/layout/activity_main.xml

@@ -8,6 +8,7 @@
         android:layout_width="match_parent"
         android:layout_height="wrap_content"
         android:onClick="toPlayActivity"
+
         android:text="播放" />
 
     <androidx.appcompat.widget.AppCompatButton
@@ -16,12 +17,6 @@
         android:onClick="toPushActivity"
         android:text="推流" />
 
-    <androidx.appcompat.widget.AppCompatButton
-        android:layout_width="match_parent"
-        android:layout_height="wrap_content"
-        android:onClick="toDataChannelActivity"
-        android:text="DataChannel"
-        android:textAllCaps="false" />
 
 
 </androidx.appcompat.widget.LinearLayoutCompat>

webrtc_player/android/app/src/main/res/layout/activity_play.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=".PlayDemoActivity">
-
-    <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/activity_player.xml

@@ -5,16 +5,28 @@
     android:layout_width="match_parent"
     android:layout_height="match_parent">
 
-    <org.webrtc.SurfaceViewRenderer
+    <com.rtc.core.RTCSurfaceView
-        android:id="@+id/surface_view_renderer"
+        android:id="@+id/surface_view_render"
-        android:layout_width="wrap_content"
+        android:layout_width="match_parent"
         android:layout_height="240dp" />
 
+    <androidx.appcompat.widget.AppCompatTextView
+        android:id="@+id/tv_status"
+        android:layout_width="wrap_content"
+        android:layout_height="wrap_content"
+        android:text=""
+        android:textSize="12sp"
+        android:textColor="@android:color/holo_red_light"
+        android:layout_alignParentRight="true"
+        android:layout_margin="6dp"
+        android:background="@color/black"
+        />
+
 
     <LinearLayout
         android:layout_width="match_parent"
         android:layout_height="wrap_content"
-        android:layout_below="@id/surface_view_renderer"
+        android:layout_below="@id/surface_view_render"
         android:orientation="vertical">
 
         <LinearLayout
@@ -41,7 +53,7 @@
         <LinearLayout
             android:layout_width="match_parent"
             android:layout_height="wrap_content"
-            android:layout_below="@id/surface_view_renderer">
+            android:layout_below="@id/surface_view_render">
 
             <androidx.appcompat.widget.AppCompatTextView
                 android:layout_width="0dp"
@@ -56,7 +68,7 @@
                 android:layout_width="0dp"
                 android:layout_height="wrap_content"
                 android:layout_weight="4"
-                android:text="li"
+                android:text="test"
                 android:gravity="center"/>
         </LinearLayout>
 

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

@@ -1,11 +1,11 @@
 <?xml version="1.0" encoding="utf-8"?>
-<androidx.constraintlayout.widget.ConstraintLayout xmlns:android="http://schemas.android.com/apk/res/android"
+<RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android"
     android:layout_width="match_parent"
     android:layout_height="match_parent"
     xmlns:app="http://schemas.android.com/apk/res-auto">
 
-    <org.webrtc.SurfaceViewRenderer
+    <com.rtc.core.RTCSurfaceView
-        android:id="@+id/surface_view_renderer"
+        android:id="@+id/surface_view_render"
         android:layout_width="match_parent"
         android:layout_height="240dp" />
 
@@ -13,8 +13,8 @@
         android:layout_width="match_parent"
         android:layout_height="wrap_content"
         android:orientation="vertical"
-        app:layout_constraintTop_toBottomOf="@id/surface_view_renderer"
+        android:layout_below="@id/surface_view_render"
-        >
+        app:layout_constraintTop_toBottomOf="@id/surface_view_render">
 
         <LinearLayout
             android:layout_width="match_parent"
@@ -40,7 +40,7 @@
         <LinearLayout
             android:layout_width="match_parent"
             android:layout_height="wrap_content"
-            android:layout_below="@id/surface_view_renderer">
+            android:layout_below="@id/surface_view_render">
 
             <androidx.appcompat.widget.AppCompatTextView
                 android:layout_width="0dp"
@@ -55,7 +55,7 @@
                 android:layout_width="0dp"
                 android:layout_height="wrap_content"
                 android:layout_weight="4"
-                android:text="li"
+                android:text="test"
                 android:gravity="center"/>
         </LinearLayout>
 
@@ -82,5 +82,14 @@
             android:textAllCaps="false"
             android:onClick="onPushFile"/>
 
+
+
+        <androidx.appcompat.widget.AppCompatButton
+            android:layout_width="match_parent"
+            android:layout_height="wrap_content"
+            android:text="Stop Push"
+            android:textAllCaps="false"
+            android:onClick="onStopPush"/>
+
     </androidx.appcompat.widget.LinearLayoutCompat>
-</androidx.constraintlayout.widget.ConstraintLayout>
+</RelativeLayout>

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

@@ -2,4 +2,5 @@
 <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" />
+    <monochrome android:drawable="@drawable/ic_launcher_foreground" />
 </adaptive-icon>

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

@@ -2,4 +2,5 @@
 <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" />
+    <monochrome android:drawable="@drawable/ic_launcher_foreground" />
 </adaptive-icon>

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

@@ -1,16 +1,7 @@
 <resources xmlns:tools="http://schemas.android.com/tools">
     <!-- Base application theme. -->
-    <style name="Theme.AndroidWebRTC" parent="Theme.MaterialComponents.DayNight.DarkActionBar">
+    <style name="Base.Theme.RTCPlayer" parent="Theme.Material3.DayNight.NoActionBar">
-        <!-- Primary brand color. -->
+        <!-- Customize your dark theme here. -->
-        <item name="colorPrimary">@color/purple_200</item>
+        <!-- <item name="colorPrimary">@color/my_dark_primary</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 +1,5 @@
 <?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 +1,3 @@
 <resources>
-    <string name="app_name">ZLMediakit WebRTC</string>
+    <string name="app_name">RTCPlayer</string>
 </resources>

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

@@ -1,16 +1,9 @@
 <resources xmlns:tools="http://schemas.android.com/tools">
     <!-- Base application theme. -->
-    <style name="Theme.AndroidWebRTC" parent="Theme.MaterialComponents.DayNight.DarkActionBar">
+    <style name="Base.Theme.RTCPlayer" parent="Theme.MaterialComponents.DayNight.DarkActionBar">
-        <!-- Primary brand color. -->
+        <!-- Customize your light theme here. -->
-        <item name="colorPrimary">@color/purple_500</item>
+        <!-- <item name="colorPrimary">@color/my_light_primary</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>
+
+    <style name="Theme.RTCPlayer" parent="Base.Theme.RTCPlayer" />
 </resources>

webrtc_player/android/app/src/test/java/com/top/player/ExampleUnitTest.java

@@ -0,0 +1,17 @@
+package com.top.player;
+
+import org.junit.Test;
+
+import static org.junit.Assert.*;
+
+/**
+ * Example local unit test, which will execute on the development machine (host).
+ *
+ * @see <a href="http://d.android.com/tools/testing">Testing documentation</a>
+ */
+public class ExampleUnitTest {
+    @Test
+    public void addition_isCorrect() {
+        assertEquals(4, 2 + 2);
+    }
+}

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/build.gradle.kts

@@ -0,0 +1,6 @@
+// Top-level build file where you can add configuration options common to all sub-projects/modules.
+plugins {
+    id("com.android.application") version "8.2.2" apply false
+    id("com.android.library") version "8.2.2" apply false
+    id("org.jetbrains.kotlin.android") version "1.9.22" apply false
+}

webrtc_player/android/demo.sdp

@@ -1,142 +0,0 @@
-v=0
-o=- 6282534540641525368 2 IN IP4 127.0.0.1
-s=-
-t=0 0
-a=group:BUNDLE 0 1 2
-a=extmap-allow-mixed
-a=msid-semantic: WMS
-m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 102 0 8 13 110 126
-c=IN IP4 0.0.0.0
-a=rtcp:9 IN IP4 0.0.0.0
-a=ice-ufrag:H2Ms
-a=ice-pwd:14ODtTVh5+lT7W0+g9EAlE4j
-a=ice-options:trickle renomination
-a=fingerprint:sha-256 47:81:90:07:43:95:D7:F2:DA:60:DA:79:E5:88:26:65:29:4A:26:28:7A:B7:AB:D3:DB:CE:C9:09:39:EB:31:91
-a=setup:actpass
-a=mid:0
-a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
-a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
-a=sendrecv
-a=msid:- ef1bcbd2-7e69-40d3-9532-0f4a698022a7
-a=rtcp-mux
-a=rtpmap:111 opus/48000/2
-a=rtcp-fb:111 transport-cc
-a=fmtp:111 minptime=10;useinbandfec=1
-a=rtpmap:63 red/48000/2
-a=fmtp:63 111/111
-a=rtpmap:9 G722/8000
-a=rtpmap:102 ILBC/8000
-a=rtpmap:0 PCMU/8000
-a=rtpmap:8 PCMA/8000
-a=rtpmap:13 CN/8000
-a=rtpmap:110 telephone-event/48000
-a=rtpmap:126 telephone-event/8000
-a=ssrc:2860285610 cname:2EW1hNfQMKxwbRrt
-a=ssrc:2860285610 msid:- ef1bcbd2-7e69-40d3-9532-0f4a698022a7
-m=video 9 UDP/TLS/RTP/SAVPF 96 97 39 40 98 99 127 103 104 105 106 107 108
-c=IN IP4 0.0.0.0
-a=rtcp:9 IN IP4 0.0.0.0
-a=ice-ufrag:H2Ms
-a=ice-pwd:14ODtTVh5+lT7W0+g9EAlE4j
-a=ice-options:trickle renomination
-a=fingerprint:sha-256 47:81:90:07:43:95:D7:F2:DA:60:DA:79:E5:88:26:65:29:4A:26:28:7A:B7:AB:D3:DB:CE:C9:09:39:EB:31:91
-a=setup:actpass
-a=mid:1
-a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
-a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-a=extmap:13 urn:3gpp:video-orientation
-a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
-a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
-a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
-a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
-a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
-a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
-a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
-a=sendonly
-a=msid:- 8b57ef63-835d-4acf-bce9-18e26cebf9b1
-a=rtcp-mux
-a=rtcp-rsize
-a=rtpmap:96 VP8/90000
-a=rtcp-fb:96 goog-remb
-a=rtcp-fb:96 transport-cc
-a=rtcp-fb:96 ccm fir
-a=rtcp-fb:96 nack
-a=rtcp-fb:96 nack pli
-a=rtpmap:97 rtx/90000
-a=fmtp:97 apt=96
-a=rtpmap:39 AV1/90000
-a=rtcp-fb:39 goog-remb
-a=rtcp-fb:39 transport-cc
-a=rtcp-fb:39 ccm fir
-a=rtcp-fb:39 nack
-a=rtcp-fb:39 nack pli
-a=rtpmap:40 rtx/90000
-a=fmtp:40 apt=39
-a=rtpmap:98 VP9/90000
-a=rtcp-fb:98 goog-remb
-a=rtcp-fb:98 transport-cc
-a=rtcp-fb:98 ccm fir
-a=rtcp-fb:98 nack
-a=rtcp-fb:98 nack pli
-a=fmtp:98 profile-id=0
-a=rtpmap:99 rtx/90000
-a=fmtp:99 apt=98
-a=rtpmap:127 H264/90000
-a=rtcp-fb:127 goog-remb
-a=rtcp-fb:127 transport-cc
-a=rtcp-fb:127 ccm fir
-a=rtcp-fb:127 nack
-a=rtcp-fb:127 nack pli
-a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
-a=rtpmap:103 rtx/90000
-a=fmtp:103 apt=127
-a=rtpmap:104 H265/90000
-a=rtcp-fb:104 goog-remb
-a=rtcp-fb:104 transport-cc
-a=rtcp-fb:104 ccm fir
-a=rtcp-fb:104 nack
-a=rtcp-fb:104 nack pli
-a=rtpmap:105 rtx/90000
-a=fmtp:105 apt=104
-a=rtpmap:106 red/90000
-a=rtpmap:107 rtx/90000
-a=fmtp:107 apt=106
-a=rtpmap:108 ulpfec/90000
-a=ssrc-group:FID 100166069 2038394461
-a=ssrc:100166069 cname:2EW1hNfQMKxwbRrt
-a=ssrc:100166069 msid:- 8b57ef63-835d-4acf-bce9-18e26cebf9b1
-a=ssrc:2038394461 cname:2EW1hNfQMKxwbRrt
-a=ssrc:2038394461 msid:- 8b57ef63-835d-4acf-bce9-18e26cebf9b1
-m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 102 0 8 13 110 126
-c=IN IP4 0.0.0.0
-a=rtcp:9 IN IP4 0.0.0.0
-a=ice-ufrag:H2Ms
-a=ice-pwd:14ODtTVh5+lT7W0+g9EAlE4j
-a=ice-options:trickle renomination
-a=fingerprint:sha-256 47:81:90:07:43:95:D7:F2:DA:60:DA:79:E5:88:26:65:29:4A:26:28:7A:B7:AB:D3:DB:CE:C9:09:39:EB:31:91
-a=setup:actpass
-a=mid:2
-a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
-a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
-a=sendonly
-a=msid:- 49d08cb1-a325-4a5f-805a-72a215eb67af
-a=rtcp-mux
-a=rtpmap:111 opus/48000/2
-a=rtcp-fb:111 transport-cc
-a=fmtp:111 minptime=10;useinbandfec=1
-a=rtpmap:63 red/48000/2
-a=fmtp:63 111/111
-a=rtpmap:9 G722/8000
-a=rtpmap:102 ILBC/8000
-a=rtpmap:0 PCMU/8000
-a=rtpmap:8 PCMA/8000
-a=rtpmap:13 CN/8000
-a=rtpmap:110 telephone-event/48000
-a=rtpmap:126 telephone-event/8000
-a=ssrc:2666757999 cname:2EW1hNfQMKxwbRrt
-a=ssrc:2666757999 msid:- 49d08cb1-a325-4a5f-805a-72a215eb67af

webrtc_player/android/gradle.properties

@@ -12,11 +12,9 @@ org.gradle.jvmargs=-Xmx2048m -Dfile.encoding=UTF-8
 # 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
+# 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

webrtc_player/android/gradle/wrapper/gradle-wrapper.properties

@@ -1,6 +1,6 @@
-#Mon Sep 19 22:08:36 CST 2022
+#Fri May 31 15:05:44 CST 2024
 distributionBase=GRADLE_USER_HOME
-distributionUrl=https\://services.gradle.org/distributions/gradle-7.3.3-bin.zip
 distributionPath=wrapper/dists
-zipStorePath=wrapper/dists
+distributionUrl=https\://services.gradle.org/distributions/gradle-8.2-bin.zip
 zipStoreBase=GRADLE_USER_HOME
+zipStorePath=wrapper/dists

webrtc_player/android/play.sdp

@@ -1,76 +0,0 @@
-v=0
-o=- 5986549880880292178 2 IN IP4 108.181.24.73
-s=-
-t=0 0
-a=group:BUNDLE 0 1
-a=msid-semantic: WMS ARDAMS
-a=ice-lite
-m=audio 8000 UDP/TLS/RTP/SAVPF 111
-c=IN IP4 108.181.24.73
-a=rtcp:8000 IN IP4 108.181.24.73
-a=ice-ufrag:bLUYSR9AH0A=_2105
-a=ice-pwd:wstnRIopRDwBZyv56AZlGuOm
-a=ice-options:trickle
-a=fingerprint:sha-256 F4:C9:0C:ED:05:93:C0:F9:AC:34:71:60:2D:3E:D0:47:91:CC:FD:38:C7:63:4A:93:C3:48:0C:B4:6F:32:76:FE
-a=setup:passive
-a=mid:0
-a=ice-lite
-a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
-a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-a=inactive
-a=rtcp-mux
-a=rtpmap:111 opus/48000/2
-a=rtcp-fb:111 transport-cc
-a=fmtp:111 minptime=10;useinbandfec=1
-a=msid:zlmediakit-mslabel zlmediakit-label-0
-a=ssrc:2 cname:zlmediakit-rtp
-a=ssrc:2 msid:zlmediakit-mslabel zlmediakit-label-0
-a=ssrc:2 mslabel:zlmediakit-mslabel
-a=ssrc:2 label:zlmediakit-label-0
-a=candidate:udpcandidate 1 udp 110 108.181.24.73 8000 typ host
-a=candidate:tcpcandidate 1 tcp 105 108.181.24.73 8000 typ host tcptype passive
-
-
-m=video 8000 UDP/TLS/RTP/SAVPF 127 103
-c=IN IP4 108.181.24.73
-a=rtcp:8000 IN IP4 108.181.24.73
-a=ice-ufrag:bLUYSR9AH0A=_2105
-a=ice-pwd:wstnRIopRDwBZyv56AZlGuOm
-a=ice-options:trickle
-a=fingerprint:sha-256 F4:C9:0C:ED:05:93:C0:F9:AC:34:71:60:2D:3E:D0:47:91:CC:FD:38:C7:63:4A:93:C3:48:0C:B4:6F:32:76:FE
-a=setup:passive
-a=mid:1
-a=ice-lite
-a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
-a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
-a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
-a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
-a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
-a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
-a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
-a=sendonly
-a=rtcp-mux
-a=rtpmap:127 H264/90000
-a=rtcp-fb:127 ccm fir
-a=rtcp-fb:127 goog-remb
-a=rtcp-fb:127 nack
-a=rtcp-fb:127 nack pli
-a=rtcp-fb:127 transport-cc
-a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
-a=rtpmap:103 rtx/90000
-a=fmtp:103 apt=127
-a=msid:zlmediakit-mslabel zlmediakit-label-1
-a=ssrc:1 cname:zlmediakit-rtp
-a=ssrc:1 msid:zlmediakit-mslabel zlmediakit-label-1
-a=ssrc:1 mslabel:zlmediakit-mslabel
-a=ssrc:1 label:zlmediakit-label-1
-a=ssrc:3 cname:zlmediakit-rtp
-a=ssrc:3 msid:zlmediakit-mslabel zlmediakit-label-1
-a=ssrc:3 mslabel:zlmediakit-mslabel
-a=ssrc:3 label:zlmediakit-label-1
-a=ssrc-group:FID 1 3
-a=candidate:udpcandidate 1 udp 110 108.181.24.73 8000 typ host
-a=candidate:tcpcandidate 1 tcp 105 108.181.24.73 8000 typ host tcptype passive

webrtc_player/android/settings.gradle

@@ -1,25 +0,0 @@
-pluginManagement {
-    repositories {
-        gradlePluginPortal()
-        google()
-        mavenCentral()
-        jcenter() // Warning: this repository is going to shut down soon
-
-        maven { url 'https://jitpack.io' }
-
-    }
-}
-dependencyResolutionManagement {
-    repositoriesMode.set(RepositoriesMode.FAIL_ON_PROJECT_REPOS)
-    repositories {
-        google()
-        mavenCentral()
-        jcenter() // Warning: this repository is going to shut down soon
-
-        maven { url 'https://jitpack.io' }
-
-    }
-}
-rootProject.name = "android"
-include ':app'
-include ':zlm'

webrtc_player/android/settings.gradle.kts

@@ -0,0 +1,25 @@
+pluginManagement {
+    repositories {
+        mavenCentral()
+        gradlePluginPortal()
+        google()
+    }
+}
+dependencyResolutionManagement {
+    repositoriesMode.set(RepositoriesMode.FAIL_ON_PROJECT_REPOS)
+    repositories {
+        mavenCentral()
+        google()
+        maven {
+            credentials {
+                username = ("6256cd6c7e8dbc28d896a661")
+                password = ("KRuEgA3WYUVy")
+            }
+            url = uri("https://packages.aliyun.com/maven/repository/2302596-release-mpvXBR/")
+        }
+    }
+}
+
+rootProject.name = "RTCPlayer"
+include(":app")
+//include(":RTCPlayer")

webrtc_player/android/success.sdp

@@ -1,112 +0,0 @@
-v=0
-o=- 6005790964057401669 2 IN IP4 127.0.0.1
-s=-
-t=0 0
-a=group:BUNDLE 0 1
-a=extmap-allow-mixed
-a=msid-semantic: WMS ARDAMS
-m=video 9 UDP/TLS/RTP/SAVPF 96 97 39 40 98 99 127 103 104 105 106 107 108
-c=IN IP4 0.0.0.0
-a=rtcp:9 IN IP4 0.0.0.0
-a=ice-ufrag:UITA
-a=ice-pwd:obg6A6ZHKhr5CGzmmgRfTk/y
-a=ice-options:trickle renomination
-a=fingerprint:sha-256 5E:23:E2:B2:0F:ED:FC:A2:20:06:24:0F:6A:4B:99:33:91:39:A7:40:A5:20:88:77:D6:99:04:FB:C7:CD:7F:33
-a=setup:actpass
-a=mid:0
-a=extmap:1 urn:ietf:params:rtp-hdrext:toffset
-a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-a=extmap:3 urn:3gpp:video-orientation
-a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
-a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
-a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
-a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
-a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
-a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
-a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
-a=sendrecv
-a=msid:ARDAMS ARDAMSv0
-a=rtcp-mux
-a=rtcp-rsize
-a=rtpmap:96 VP8/90000
-a=rtcp-fb:96 goog-remb
-a=rtcp-fb:96 transport-cc
-a=rtcp-fb:96 ccm fir
-a=rtcp-fb:96 nack
-a=rtcp-fb:96 nack pli
-a=rtpmap:97 rtx/90000
-a=fmtp:97 apt=96
-a=rtpmap:39 AV1/90000
-a=rtcp-fb:39 goog-remb
-a=rtcp-fb:39 transport-cc
-a=rtcp-fb:39 ccm fir
-a=rtcp-fb:39 nack
-a=rtcp-fb:39 nack pli
-a=rtpmap:40 rtx/90000
-a=fmtp:40 apt=39
-a=rtpmap:98 VP9/90000
-a=rtcp-fb:98 goog-remb
-a=rtcp-fb:98 transport-cc
-a=rtcp-fb:98 ccm fir
-a=rtcp-fb:98 nack
-a=rtcp-fb:98 nack pli
-a=fmtp:98 profile-id=0
-a=rtpmap:99 rtx/90000
-a=fmtp:99 apt=98
-a=rtpmap:127 H264/90000
-a=rtcp-fb:127 goog-remb
-a=rtcp-fb:127 transport-cc
-a=rtcp-fb:127 ccm fir
-a=rtcp-fb:127 nack
-a=rtcp-fb:127 nack pli
-a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
-a=rtpmap:103 rtx/90000
-a=fmtp:103 apt=127
-a=rtpmap:104 H265/90000
-a=rtcp-fb:104 goog-remb
-a=rtcp-fb:104 transport-cc
-a=rtcp-fb:104 ccm fir
-a=rtcp-fb:104 nack
-a=rtcp-fb:104 nack pli
-a=rtpmap:105 rtx/90000
-a=fmtp:105 apt=104
-a=rtpmap:106 red/90000
-a=rtpmap:107 rtx/90000
-a=fmtp:107 apt=106
-a=rtpmap:108 ulpfec/90000
-a=ssrc-group:FID 1882091013 3163545249
-a=ssrc:1882091013 cname:L6diPFsW7sR8uU/a
-a=ssrc:1882091013 msid:ARDAMS ARDAMSv0
-a=ssrc:3163545249 cname:L6diPFsW7sR8uU/a
-a=ssrc:3163545249 msid:ARDAMS ARDAMSv0
-m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 102 0 8 13 110 126
-c=IN IP4 0.0.0.0
-a=rtcp:9 IN IP4 0.0.0.0
-a=ice-ufrag:UITA
-a=ice-pwd:obg6A6ZHKhr5CGzmmgRfTk/y
-a=ice-options:trickle renomination
-a=fingerprint:sha-256 5E:23:E2:B2:0F:ED:FC:A2:20:06:24:0F:6A:4B:99:33:91:39:A7:40:A5:20:88:77:D6:99:04:FB:C7:CD:7F:33
-a=setup:actpass
-a=mid:1
-a=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level
-a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
-a=sendrecv
-a=msid:ARDAMS ARDAMSa0
-a=rtcp-mux
-a=rtpmap:111 opus/48000/2
-a=rtcp-fb:111 transport-cc
-a=fmtp:111 minptime=10;useinbandfec=1
-a=rtpmap:63 red/48000/2
-a=fmtp:63 111/111
-a=rtpmap:9 G722/8000
-a=rtpmap:102 ILBC/8000
-a=rtpmap:0 PCMU/8000
-a=rtpmap:8 PCMA/8000
-a=rtpmap:13 CN/8000
-a=rtpmap:110 telephone-event/48000
-a=rtpmap:126 telephone-event/8000
-a=ssrc:1776947399 cname:L6diPFsW7sR8uU/a
-a=ssrc:1776947399 msid:ARDAMS ARDAMSa0

webrtc_player/android/this.sdp

@@ -1,119 +0,0 @@
-v=0
-o=- 2629706501878304631 2 IN IP4 127.0.0.1
-s=-
-t=0 0
-a=group:BUNDLE 0 1
-a=extmap-allow-mixed
-a=msid-semantic: WMS ARDAMS
-m=video 9 UDP/TLS/RTP/SAVPF 96 97 39 40 98 99 127 103 104 105 106 107 108 43
-c=IN IP4 0.0.0.0
-a=rtcp:9 IN IP4 0.0.0.0
-a=ice-ufrag:+zOA
-a=ice-pwd:q88Il7adjqdu6SeSqAdsv8lv
-a=ice-options:trickle renomination
-a=fingerprint:sha-256 8D:0B:86:0B:8C:DF:22:99:85:DD:60:41:AA:FE:F1:98:FE:FB:29:32:5C:11:6E:47:54:82:21:6C:A0:4F:60:B2
-a=setup:actpass
-a=mid:0
-a=extmap:1 urn:ietf:params:rtp-hdrext:toffset
-a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-a=extmap:3 urn:3gpp:video-orientation
-a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
-a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
-a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
-a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
-a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
-a=extmap:10 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
-a=extmap:11 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
-a=sendrecv
-a=msid:ARDAMS ARDAMSv0
-a=rtcp-mux
-a=rtcp-rsize
-a=rtpmap:96 VP8/90000
-a=rtcp-fb:96 goog-remb
-a=rtcp-fb:96 transport-cc
-a=rtcp-fb:96 ccm fir
-a=rtcp-fb:96 nack
-a=rtcp-fb:96 nack pli
-a=rtpmap:97 rtx/90000
-a=fmtp:97 apt=96
-a=rtpmap:39 AV1/90000
-a=rtcp-fb:39 goog-remb
-a=rtcp-fb:39 transport-cc
-a=rtcp-fb:39 ccm fir
-a=rtcp-fb:39 nack
-a=rtcp-fb:39 nack pli
-a=rtpmap:40 rtx/90000
-a=fmtp:40 apt=39
-a=rtpmap:98 VP9/90000
-a=rtcp-fb:98 goog-remb
-a=rtcp-fb:98 transport-cc
-a=rtcp-fb:98 ccm fir
-a=rtcp-fb:98 nack
-a=rtcp-fb:98 nack pli
-a=fmtp:98 profile-id=0
-a=rtpmap:99 rtx/90000
-a=fmtp:99 apt=98
-a=rtpmap:127 H264/90000
-a=rtcp-fb:127 goog-remb
-a=rtcp-fb:127 transport-cc
-a=rtcp-fb:127 ccm fir
-a=rtcp-fb:127 nack
-a=rtcp-fb:127 nack pli
-a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
-a=rtpmap:103 rtx/90000
-a=fmtp:103 apt=127
-a=rtpmap:104 H265/90000
-a=rtcp-fb:104 goog-remb
-a=rtcp-fb:104 transport-cc
-a=rtcp-fb:104 ccm fir
-a=rtcp-fb:104 nack
-a=rtcp-fb:104 nack pli
-a=rtpmap:105 rtx/90000
-a=fmtp:105 apt=104
-a=rtpmap:106 red/90000
-a=rtpmap:107 rtx/90000
-a=fmtp:107 apt=106
-a=rtpmap:108 ulpfec/90000
-a=rtpmap:43 flexfec-03/90000
-a=rtcp-fb:43 goog-remb
-a=rtcp-fb:43 transport-cc
-a=fmtp:43 repair-window=10000000
-a=ssrc-group:FID 129065984 1082217256
-a=ssrc-group:FEC-FR 129065984 4071693473
-a=ssrc:129065984 cname:T1fpJkmBNuVUtXM8
-a=ssrc:129065984 msid:ARDAMS ARDAMSv0
-a=ssrc:1082217256 cname:T1fpJkmBNuVUtXM8
-a=ssrc:1082217256 msid:ARDAMS ARDAMSv0
-a=ssrc:4071693473 cname:T1fpJkmBNuVUtXM8
-a=ssrc:4071693473 msid:ARDAMS ARDAMSv0
-m=audio 9 UDP/TLS/RTP/SAVPF 111 63 9 102 0 8 13 110 126
-c=IN IP4 0.0.0.0
-a=rtcp:9 IN IP4 0.0.0.0
-a=ice-ufrag:+zOA
-a=ice-pwd:q88Il7adjqdu6SeSqAdsv8lv
-a=ice-options:trickle renomination
-a=fingerprint:sha-256 8D:0B:86:0B:8C:DF:22:99:85:DD:60:41:AA:FE:F1:98:FE:FB:29:32:5C:11:6E:47:54:82:21:6C:A0:4F:60:B2
-a=setup:actpass
-a=mid:1
-a=extmap:14 urn:ietf:params:rtp-hdrext:ssrc-audio-level
-a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
-a=extmap:4 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
-a=extmap:9 urn:ietf:params:rtp-hdrext:sdes:mid
-a=sendrecv
-a=msid:ARDAMS ARDAMSa0
-a=rtcp-mux
-a=rtpmap:111 opus/48000/2
-a=rtcp-fb:111 transport-cc
-a=fmtp:111 minptime=10;useinbandfec=1
-a=rtpmap:63 red/48000/2
-a=fmtp:63 111/111
-a=rtpmap:9 G722/8000
-a=rtpmap:102 ILBC/8000
-a=rtpmap:0 PCMU/8000
-a=rtpmap:8 PCMA/8000
-a=rtpmap:13 CN/8000
-a=rtpmap:110 telephone-event/48000
-a=rtpmap:126 telephone-event/8000
-a=ssrc:2019904101 cname:T1fpJkmBNuVUtXM8
-a=ssrc:2019904101 msid:ARDAMS ARDAMSa0

webrtc_player/android/zlm/.gitignore

@@ -1 +0,0 @@
-/build

webrtc_player/android/zlm/build.gradle

@@ -1,61 +0,0 @@
-plugins {
-    id 'com.android.library'
-    id 'org.jetbrains.kotlin.android'
-}
-
-android {
-    namespace 'com.zlm.rtc'
-    compileSdk 34
-
-    defaultConfig {
-        minSdk 24
-
-        testInstrumentationRunner "androidx.test.runner.AndroidJUnitRunner"
-        consumerProguardFiles "consumer-rules.pro"
-        externalNativeBuild {
-            cmake {
-                cppFlags ""
-            }
-        }
-    }
-
-    buildTypes {
-        release {
-            minifyEnabled false
-            proguardFiles getDefaultProguardFile('proguard-android-optimize.txt'), 'proguard-rules.pro'
-        }
-    }
-    externalNativeBuild {
-        cmake {
-            path "src/main/cpp/CMakeLists.txt"
-            version "3.22.1"
-        }
-    }
-    compileOptions {
-        sourceCompatibility JavaVersion.VERSION_1_8
-        targetCompatibility JavaVersion.VERSION_1_8
-    }
-    kotlinOptions {
-        jvmTarget = '1.8'
-    }
-    sourceSets {
-        main{
-            jniLibs.srcDirs =  ['libs']
-        }
-    }
-}
-
-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"
-}

webrtc_player/android/zlm/libs/arm64-v8a/libZLToolKit.a

@@ -1,3 +0,0 @@
-version https://git-lfs.github.com/spec/v1
-oid sha256:bc218b4b68589b49cb72278af6759deea0146c47079eea708bba5669545e9005
-size 183748898

webrtc_player/android/zlm/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/zlm/src/androidTest/java/com/zlm/rtc/ExampleInstrumentedTest.kt

@@ -1,24 +0,0 @@
-package com.zlm.rtc
-
-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.zlm.rtc.test", appContext.packageName)
-    }
-}

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

@@ -1,23 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<manifest xmlns:android="http://schemas.android.com/apk/res/android">
-
-    <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"/>
-
-    <uses-permission android:name="android.permission.FOREGROUND_SERVICE"/>
-
-    <application>
-        <service android:name="com.zlm.rtc.push.ScreenShareService"
-            android:enabled="true"
-            android:exported="true"
-            android:foregroundServiceType="mediaProjection"/>
-    </application>
-</manifest>

webrtc_player/android/zlm/src/main/cpp/CMakeLists.txt

@@ -1,46 +0,0 @@
-# For more information about using CMake with Android Studio, read the
-# documentation: https://d.android.com/studio/projects/add-native-code.html.
-# For more examples on how to use CMake, see https://github.com/android/ndk-samples.
-
-# Sets the minimum CMake version required for this project.
-cmake_minimum_required(VERSION 3.22.1)
-
-# Declares the project name. The project name can be accessed via ${ PROJECT_NAME},
-# Since this is the top level CMakeLists.txt, the project name is also accessible
-# with ${CMAKE_PROJECT_NAME} (both CMake variables are in-sync within the top level
-# build script scope).
-project("rtc")
-
-
-
-add_library(ZLToolKit IMPORTED STATIC)
-set_target_properties(ZLToolKit PROPERTIES
-        IMPORTED_LOCATION "${CMAKE_CURRENT_SOURCE_DIR}/../../../libs/${CMAKE_ANDROID_ARCH_ABI}/libZLToolKit.a"
-        INTERFACE_INCLUDE_DIRECTORIES "${CMAKE_CURRENT_SOURCE_DIR}/include"
-        )
-
-# Creates and names a library, sets it as either STATIC
-# or SHARED, and provides the relative paths to its source code.
-# You can define multiple libraries, and CMake builds them for you.
-# Gradle automatically packages shared libraries with your APK.
-#
-# In this top level CMakeLists.txt, ${CMAKE_PROJECT_NAME} is used to define
-# the target library name; in the sub-module's CMakeLists.txt, ${PROJECT_NAME}
-# is preferred for the same purpose.
-#
-# In order to load a library into your app from Java/Kotlin, you must call
-# System.loadLibrary() and pass the name of the library defined here;
-# for GameActivity/NativeActivity derived applications, the same library name must be
-# used in the AndroidManifest.xml file.
-add_library(${CMAKE_PROJECT_NAME} SHARED
-        # List C/C++ source files with relative paths to this CMakeLists.txt.
-        rtc.cpp)
-
-# Specifies libraries CMake should link to your target library. You
-# can link libraries from various origins, such as libraries defined in this
-# build script, prebuilt third-party libraries, or Android system libraries.
-target_link_libraries(${CMAKE_PROJECT_NAME}
-        # List libraries link to the target library
-        android
-        log
-        ZLToolKit)

webrtc_player/android/zlm/src/main/cpp/include/Codec/AACEncoder.h

@@ -1,38 +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 CODEC_AACENCODER_H_
-#define CODEC_AACENCODER_H_
-
-namespace mediakit {
-
-class AACEncoder {
-public:
-    AACEncoder(void);
-    virtual ~AACEncoder(void);
-    bool init(int iSampleRate, int iAudioChannel, int iAudioSampleBit);
-    int inputData(char *pcData, int iLen, unsigned char **ppucOutBuffer);
-
-private:
-    unsigned char *_pucPcmBuf = nullptr;
-    unsigned int _uiPcmLen = 0;
-
-    unsigned char *_pucAacBuf = nullptr;
-    void *_hEncoder = nullptr;
-
-    unsigned long _ulInputSamples = 0;
-    unsigned long _ulMaxInputBytes = 0;
-    unsigned long _ulMaxOutputBytes = 0;
-
-};
-
-} /* namespace mediakit */
-
-#endif /* CODEC_AACENCODER_H_ */

webrtc_player/android/zlm/src/main/cpp/include/Codec/H264Encoder.h

@@ -1,47 +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 CODEC_H264ENCODER_H_
-#define CODEC_H264ENCODER_H_
-
-#include <cstdint>
-#ifdef __cplusplus
-extern "C" {
-#endif //__cplusplus
-#include <x264.h>
-#ifdef __cplusplus
-}
-#endif //__cplusplus
-
-namespace mediakit {
-
-class H264Encoder {
-public:
-    typedef struct {
-        int iType;
-        int iLength;
-        uint8_t *pucData;
-    } H264Frame;
-
-    H264Encoder();
-    ~H264Encoder();
-
-    bool init(int iWidth, int iHeight, int iFps, int iBitRate);
-    int inputData(char *yuv[3], int linesize[3], int64_t cts, H264Frame **out_frame);
-
-private:
-    x264_t *_pX264Handle = nullptr;
-    x264_picture_t *_pPicIn = nullptr;
-    x264_picture_t *_pPicOut = nullptr;
-    H264Frame _aFrames[10];
-};
-
-} /* namespace mediakit */
-
-#endif /* CODEC_H264ENCODER_H_ */

webrtc_player/android/zlm/src/main/cpp/include/Codec/Transcode.h

@@ -1,148 +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_TRANSCODE_H
-#define ZLMEDIAKIT_TRANSCODE_H
-
-#if defined(ENABLE_FFMPEG)
-
-#include "Util/TimeTicker.h"
-#include "Common/MediaSink.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-#include "libswscale/swscale.h"
-#include "libavutil/avutil.h"
-#include "libavutil/pixdesc.h"
-#include "libavcodec/avcodec.h"
-#include "libswresample/swresample.h"
-#include "libavutil/audio_fifo.h"
-#include "libavutil/imgutils.h"
-#ifdef __cplusplus
-}
-#endif
-
-namespace mediakit {
-
-class FFmpegFrame {
-public:
-    using Ptr = std::shared_ptr<FFmpegFrame>;
-
-    FFmpegFrame(std::shared_ptr<AVFrame> frame = nullptr);
-    ~FFmpegFrame();
-
-    AVFrame *get() const;
-    void fillPicture(AVPixelFormat target_format, int target_width, int target_height);
-
-private:
-    char *_data = nullptr;
-    std::shared_ptr<AVFrame> _frame;
-};
-
-class FFmpegSwr {
-public:
-    using Ptr = std::shared_ptr<FFmpegSwr>;
-
-    FFmpegSwr(AVSampleFormat output, int channel, int channel_layout, int samplerate);
-    ~FFmpegSwr();
-    FFmpegFrame::Ptr inputFrame(const FFmpegFrame::Ptr &frame);
-
-private:
-    int _target_channels;
-    int _target_channel_layout;
-    int _target_samplerate;
-    AVSampleFormat _target_format;
-    SwrContext *_ctx = nullptr;
-};
-
-class TaskManager {
-public:
-    virtual ~TaskManager();
-
-    void setMaxTaskSize(size_t size);
-    void stopThread(bool drop_task);
-
-protected:
-    void startThread(const std::string &name);
-    bool addEncodeTask(std::function<void()> task);
-    bool addDecodeTask(bool key_frame, std::function<void()> task);
-    bool isEnabled() const;
-
-private:
-    void onThreadRun(const std::string &name);
-
-private:
-    class ThreadExitException : public std::runtime_error {
-    public:
-        ThreadExitException() : std::runtime_error("exit") {}
-    };
-
-private:
-    bool _decode_drop_start = false;
-    bool _exit = false;
-    size_t _max_task = 30;
-    std::mutex _task_mtx;
-    toolkit::semaphore _sem;
-    toolkit::List<std::function<void()> > _task;
-    std::shared_ptr<std::thread> _thread;
-};
-
-class FFmpegDecoder : public TaskManager {
-public:
-    using Ptr = std::shared_ptr<FFmpegDecoder>;
-    using onDec = std::function<void(const FFmpegFrame::Ptr &)>;
-
-    FFmpegDecoder(const Track::Ptr &track, int thread_num = 2, const std::vector<std::string> &codec_name = {});
-    ~FFmpegDecoder() override;
-
-    bool inputFrame(const Frame::Ptr &frame, bool live, bool async, bool enable_merge = true);
-    void setOnDecode(onDec cb);
-    void flush();
-    const AVCodecContext *getContext() const;
-
-private:
-    void onDecode(const FFmpegFrame::Ptr &frame);
-    bool inputFrame_l(const Frame::Ptr &frame, bool live, bool enable_merge);
-    bool decodeFrame(const char *data, size_t size, uint64_t dts, uint64_t pts, bool live, bool key_frame);
-
-private:
-    bool _do_merger = false;
-    toolkit::Ticker _ticker;
-    onDec _cb;
-    std::shared_ptr<AVCodecContext> _context;
-    FrameMerger _merger{FrameMerger::h264_prefix};
-};
-
-class FFmpegSws {
-public:
-    using Ptr = std::shared_ptr<FFmpegSws>;
-
-    FFmpegSws(AVPixelFormat output, int width, int height);
-    ~FFmpegSws();
-    FFmpegFrame::Ptr inputFrame(const FFmpegFrame::Ptr &frame);
-    int inputFrame(const FFmpegFrame::Ptr &frame, uint8_t *data);
-
-private:
-    FFmpegFrame::Ptr inputFrame(const FFmpegFrame::Ptr &frame, int &ret, uint8_t *data);
-
-private:
-    int _target_width = 0;
-    int _target_height = 0;
-    int _src_width = 0;
-    int _src_height = 0;
-    SwsContext *_ctx = nullptr;
-    AVPixelFormat _src_format = AV_PIX_FMT_NONE;
-    AVPixelFormat _target_format = AV_PIX_FMT_NONE;
-};
-
-}//namespace mediakit
-#endif// ENABLE_FFMPEG
-#endif //ZLMEDIAKIT_TRANSCODE_H

webrtc_player/android/zlm/src/main/cpp/include/Common/Device.h

@@ -1,136 +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 DEVICE_DEVICE_H_
-#define DEVICE_DEVICE_H_
-
-#include <memory>
-#include <string>
-#include <functional>
-#include "Util/TimeTicker.h"
-#include "Common/MultiMediaSourceMuxer.h"
-
-namespace mediakit {
-
-class H264Encoder;
-class AACEncoder;
-
-class VideoInfo {
-public:
-    CodecId codecId = CodecH264;
-    int iWidth;
-    int iHeight;
-    float iFrameRate;
-    int iBitRate = 2 * 1024 * 1024;
-};
-
-class AudioInfo {
-public:
-    CodecId codecId = CodecAAC;
-    int iChannel;
-    int iSampleBit;
-    int iSampleRate;
-};
-
-/**
- * MultiMediaSourceMuxer类的包装，方便初学者使用
- */
-class DevChannel  : public MultiMediaSourceMuxer{
-public:
-    using Ptr = std::shared_ptr<DevChannel>;
-
-    //fDuration<=0为直播，否则为点播
-    DevChannel(const MediaTuple& tuple, float duration = 0, const ProtocolOption &option = ProtocolOption())
-        : MultiMediaSourceMuxer(tuple, duration, option) {}
-
-    /**
-     * 初始化视频Track
-     * 相当于MultiMediaSourceMuxer::addTrack(VideoTrack::Ptr );
-     * @param info 视频相关信息
-     */
-    bool initVideo(const VideoInfo &info);
-
-    /**
-     * 初始化音频Track
-     * 相当于MultiMediaSourceMuxer::addTrack(AudioTrack::Ptr );
-     * @param info 音频相关信息
-     */
-    bool initAudio(const AudioInfo &info);
-
-    /**
-     * 输入264帧
-     * @param data 264单帧数据指针
-     * @param len 数据指针长度
-     * @param dts 解码时间戳，单位毫秒；等于0时内部会自动生成时间戳
-     * @param pts 播放时间戳，单位毫秒；等于0时内部会赋值为dts
-     */
-    bool inputH264(const char *data, int len, uint64_t dts, uint64_t pts = 0);
-
-    /**
-     * 输入265帧
-     * @param data 265单帧数据指针
-     * @param len 数据指针长度
-     * @param dts 解码时间戳，单位毫秒；等于0时内部会自动生成时间戳
-     * @param pts 播放时间戳，单位毫秒；等于0时内部会赋值为dts
-     */
-    bool inputH265(const char *data, int len, uint64_t dts, uint64_t pts = 0);
-
-    /**
-     * 输入aac帧
-     * @param data_without_adts 不带adts头的aac帧
-     * @param len 帧数据长度
-     * @param dts 时间戳，单位毫秒
-     * @param adts_header adts头
-     */
-    bool inputAAC(const char *data_without_adts, int len, uint64_t dts, const char *adts_header);
-
-    /**
-     * 输入OPUS/G711音频帧
-     * @param data 音频帧
-     * @param len 帧数据长度
-     * @param dts 时间戳，单位毫秒
-     */
-    bool inputAudio(const char *data, int len, uint64_t dts);
-
-    /**
-     * 输入yuv420p视频帧，内部会完成编码并调用inputH264方法
-     * @param yuv yuv420p数据指针
-     * @param linesize yuv420p数据linesize
-     * @param cts 采集时间戳，单位毫秒
-     */
-    bool inputYUV(char *yuv[3], int linesize[3], uint64_t cts);
-
-    /**
-     * 输入pcm数据，内部会完成编码并调用inputAAC方法
-     * @param data pcm数据指针，int16整形
-     * @param len pcm数据长度
-     * @param cts 采集时间戳，单位毫秒
-     */
-    bool inputPCM(char *data, int len, uint64_t cts);
-
-    //// 重载基类方法，确保线程安全 ////
-    bool inputFrame(const Frame::Ptr &frame) override;
-    bool addTrack(const Track::Ptr & track) override;
-    void addTrackCompleted() override;
-
-private:
-    MediaOriginType getOriginType(MediaSource &sender) const override;
-
-private:
-    std::shared_ptr<H264Encoder> _pH264Enc;
-    std::shared_ptr<AACEncoder> _pAacEnc;
-    std::shared_ptr<VideoInfo> _video;
-    std::shared_ptr<AudioInfo> _audio;
-    toolkit::SmoothTicker _aTicker[2];
-};
-
-} /* namespace mediakit */
-
-#endif /* DEVICE_DEVICE_H_ */

webrtc_player/android/zlm/src/main/cpp/include/Common/JemallocUtil.h

@@ -1,28 +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_JEMALLOCUTIL_H
-#define ZLMEDIAKIT_JEMALLOCUTIL_H
-#include <functional>
-#include <string>
-#include <cstdint>
-namespace mediakit {
-class JemallocUtil {
-public:
-    static void enable_profiling();
-
-    static void disable_profiling();
-
-    static void dump(const std::string &file_name);
-    static std::string get_malloc_stats();
-    static void some_malloc_stats(const std::function<void(const char *, uint64_t)> &fn);
-};
-} // namespace mediakit
-#endif // ZLMEDIAKIT_JEMALLOCUTIL_H

webrtc_player/android/zlm/src/main/cpp/include/Common/MediaSink.h

@@ -1,218 +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_MEDIASINK_H
-#define ZLMEDIAKIT_MEDIASINK_H
-
-#include <mutex>
-#include <memory>
-#include "Util/TimeTicker.h"
-#include "Extension/Frame.h"
-#include "Extension/Track.h"
-
-namespace mediakit{
-
-class TrackListener {
-public:
-    virtual ~TrackListener() = default;
-
-    /**
-     * 添加track，内部会调用Track的clone方法
-     * 只会克隆sps pps这些信息 ，而不会克隆Delegate相关关系
-     * @param track
-     */
-    virtual bool addTrack(const Track::Ptr & track) = 0;
-
-    /**
-     * 添加track完毕
-     */
-    virtual void addTrackCompleted() {};
-
-    /**
-     * 重置track
-     */
-    virtual void resetTracks() {};
-};
-
-class MediaSinkInterface : public FrameWriterInterface, public TrackListener {
-public:
-    using Ptr = std::shared_ptr<MediaSinkInterface>;
-};
-
-/**
- * aac静音音频添加器
- */
-class MuteAudioMaker : public FrameDispatcher {
-public:
-    using Ptr = std::shared_ptr<MuteAudioMaker>;
-    bool inputFrame(const Frame::Ptr &frame) override;
-
-private:
-    int _track_index = -1;
-    uint64_t _audio_idx = 0;
-};
-
-/**
- * 该类的作用是等待Track ready()返回true也就是就绪后再通知派生类进行下一步的操作
- * 目的是输入Frame前由Track截取处理下，以便获取有效的信息（譬如sps pps aa_cfg）
- */
-class MediaSink : public MediaSinkInterface, public TrackSource{
-public:
-    using Ptr = std::shared_ptr<MediaSink>;
-    /**
-     * 输入frame
-     * @param frame
-     */
-    bool inputFrame(const Frame::Ptr &frame) override;
-
-    /**
-     * 添加track，内部会调用Track的clone方法
-     * 只会克隆sps pps这些信息 ，而不会克隆Delegate相关关系
-     * @param track
-     */
-    bool addTrack(const Track::Ptr & track) override;
-
-    /**
-     * 添加Track完毕，如果是单Track，会最多等待3秒才会触发onAllTrackReady
-     * 这样会增加生成流的延时，如果添加了音视频双Track，那么可以不调用此方法
-     * 否则为了降低流注册延时，请手动调用此方法
-     */
-    void addTrackCompleted() override;
-
-    /**
-     * 设置最大track数，取值范围>=1；该方法与addTrackCompleted类型；
-     * 在设置单track时，可以加快媒体注册速度
-     */
-    void setMaxTrackCount(size_t i);
-
-    /**
-     * 重置track
-     */
-    void resetTracks() override;
-
-    /**
-     * 获取所有Track
-     * @param trackReady 是否获取已经准备好的Track
-     */
-    std::vector<Track::Ptr> getTracks(bool trackReady = true) const override;
-
-    /**
-     * 判断是否已经触发onAllTrackReady事件
-     */
-    bool isAllTrackReady() const;
-
-    /**
-     * 设置是否开启音频
-     */
-    void enableAudio(bool flag);
-
-    /**
-     * 设置单音频
-     */
-    void setOnlyAudio();
-
-    /**
-     * 设置是否开启添加静音音频
-     */
-    void enableMuteAudio(bool flag);
-
-    /**
-     * 是否有视频track
-     */
-    bool haveVideo() const;
-
-protected:
-    /**
-     * 某track已经准备好，其ready()状态返回true，
-     * 此时代表可以获取其例如sps pps等相关信息了
-     * @param track
-     */
-    virtual bool onTrackReady(const Track::Ptr & track) { return false; };
-
-    /**
-     * 所有Track已经准备好，
-     */
-    virtual void onAllTrackReady() {};
-
-    /**
-     * 某Track输出frame，在onAllTrackReady触发后才会调用此方法
-     * @param frame
-     */
-    virtual bool onTrackFrame(const Frame::Ptr &frame) { return false; };
-
-private:
-    /**
-     * 触发onAllTrackReady事件
-     */
-    void emitAllTrackReady();
-
-    /**
-     * 检查track是否准备完毕
-     */
-    void checkTrackIfReady();
-    void onAllTrackReady_l();
-    /**
-     * 添加aac静音轨道
-     */
-    bool addMuteAudioTrack();
-
-private:
-    bool _audio_add = false;
-    bool _have_video = false;
-    bool _enable_audio = true;
-    bool _only_audio = false;
-    bool _add_mute_audio = true;
-    bool _all_track_ready = false;
-    size_t _max_track_size = 2;
-
-    toolkit::Ticker _ticker;
-    MuteAudioMaker::Ptr _mute_audio_maker;
-
-    std::unordered_map<int, toolkit::List<Frame::Ptr> > _frame_unread;
-    std::unordered_map<int, std::function<void()> > _track_ready_callback;
-    std::unordered_map<int, std::pair<Track::Ptr, bool/*got frame*/> > _track_map;
-};
-
-
-class MediaSinkDelegate : public MediaSink {
-public:
-    /**
-     * 设置track监听器
-     */
-    void setTrackListener(TrackListener *listener);
-
-protected:
-    void resetTracks() override;
-    bool onTrackReady(const Track::Ptr & track) override;
-    void onAllTrackReady() override;
-
-private:
-    TrackListener *_listener = nullptr;
-};
-
-class Demuxer : protected TrackListener, public TrackSource {
-public:
-    void setTrackListener(TrackListener *listener, bool wait_track_ready = false);
-    std::vector<Track::Ptr> getTracks(bool trackReady = true) const override;
-
-protected:
-    bool addTrack(const Track::Ptr &track) override;
-    void addTrackCompleted() override;
-    void resetTracks() override;
-
-private:
-    MediaSink::Ptr _sink;
-    TrackListener *_listener = nullptr;
-    std::vector<Track::Ptr> _origin_track;
-};
-
-}//namespace mediakit
-
-#endif //ZLMEDIAKIT_MEDIASINK_H

webrtc_player/android/zlm/src/main/cpp/include/Common/MediaSource.h

@@ -1,443 +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_MEDIASOURCE_H
-#define ZLMEDIAKIT_MEDIASOURCE_H
-
-#include <string>
-#include <atomic>
-#include <memory>
-#include <functional>
-#include "Network/Socket.h"
-#include "Extension/Track.h"
-#include "Record/Recorder.h"
-
-namespace toolkit {
-class Session;
-} // namespace toolkit
-
-namespace mediakit {
-
-enum class MediaOriginType : uint8_t {
-    unknown = 0,
-    rtmp_push ,
-    rtsp_push,
-    rtp_push,
-    pull,
-    ffmpeg_pull,
-    mp4_vod,
-    device_chn,
-    rtc_push,
-    srt_push
-};
-
-std::string getOriginTypeString(MediaOriginType type);
-
-class MediaSource;
-class MultiMediaSourceMuxer;
-class MediaSourceEvent {
-public:
-    friend class MediaSource;
-
-    class NotImplemented : public std::runtime_error {
-    public:
-        template<typename ...T>
-        NotImplemented(T && ...args) : std::runtime_error(std::forward<T>(args)...) {}
-    };
-
-    virtual ~MediaSourceEvent() = default;
-
-    // 获取媒体源类型
-    virtual MediaOriginType getOriginType(MediaSource &sender) const { return MediaOriginType::unknown; }
-    // 获取媒体源url或者文件路径
-    virtual std::string getOriginUrl(MediaSource &sender) const;
-    // 获取媒体源客户端相关信息
-    virtual std::shared_ptr<toolkit::SockInfo> getOriginSock(MediaSource &sender) const { return nullptr; }
-
-    // 通知拖动进度条
-    virtual bool seekTo(MediaSource &sender, uint32_t stamp) { return false; }
-    // 通知暂停或恢复
-    virtual bool pause(MediaSource &sender, bool pause) { return false; }
-    // 通知倍数
-    virtual bool speed(MediaSource &sender, float speed) { return false; }
-    // 通知其停止产生流
-    virtual bool close(MediaSource &sender) { return false; }
-    // 获取观看总人数，此函数一般强制重载
-    virtual int totalReaderCount(MediaSource &sender) { throw NotImplemented(toolkit::demangle(typeid(*this).name()) + "::totalReaderCount not implemented"); }
-    // 通知观看人数变化
-    virtual void onReaderChanged(MediaSource &sender, int size);
-    //流注册或注销事件
-    virtual void onRegist(MediaSource &sender, bool regist) {}
-    // 获取丢包率
-    virtual float getLossRate(MediaSource &sender, TrackType type) { return -1; }
-    // 获取所在线程, 此函数一般强制重载
-    virtual toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) { throw NotImplemented(toolkit::demangle(typeid(*this).name()) + "::getOwnerPoller not implemented"); }
-
-    ////////////////////////仅供MultiMediaSourceMuxer对象继承////////////////////////
-    // 开启或关闭录制
-    virtual bool setupRecord(MediaSource &sender, Recorder::type type, bool start, const std::string &custom_path, size_t max_second) { return false; };
-    // 获取录制状态
-    virtual bool isRecording(MediaSource &sender, Recorder::type type) { return false; }
-    // 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; }
-
-    class SendRtpArgs {
-    public:
-        enum Type { kRtpRAW = 0, kRtpPS = 1, kRtpTS = 2 };
-        // 是否采用udp方式发送rtp
-        bool is_udp = true;
-        // rtp类型
-        Type type = kRtpPS;
-        //发送es流时指定是否只发送纯音频流
-        bool only_audio = false;
-        //tcp被动方式
-        bool passive = false;
-        // rtp payload type
-        uint8_t pt = 96;
-        //是否支持同ssrc多服务器发送
-        bool ssrc_multi_send = false;
-        // 指定rtp ssrc
-        std::string ssrc;
-        // 指定本地发送端口
-        uint16_t src_port = 0;
-        // 发送目标端口
-        uint16_t dst_port;
-        // 发送目标主机地址，可以是ip或域名
-        std::string dst_url;
-
-        //udp发送时，是否开启rr rtcp接收超时判断
-        bool udp_rtcp_timeout = false;
-        //tcp被动发送服务器延时关闭事件，单位毫秒；设置为0时，则使用默认值5000ms
-        uint32_t tcp_passive_close_delay_ms = 0;
-        //udp 发送时，rr rtcp包接收超时时间，单位毫秒
-        uint32_t rtcp_timeout_ms = 30 * 1000;
-        //udp 发送时，发送sr rtcp包间隔，单位毫秒
-        uint32_t rtcp_send_interval_ms = 5 * 1000;
-
-        //发送rtp同时接收，一般用于双向语言对讲, 如果不为空，说明开启接收
-        std::string recv_stream_id;
-    };
-
-    // 开始发送ps-rtp
-    virtual void startSendRtp(MediaSource &sender, const SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) { cb(0, toolkit::SockException(toolkit::Err_other, "not implemented"));};
-    // 停止发送ps-rtp
-    virtual bool stopSendRtp(MediaSource &sender, const std::string &ssrc) {return false; }
-
-private:
-    toolkit::Timer::Ptr _async_close_timer;
-};
-
-
-template <typename MAP, typename KEY, typename TYPE>
-static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
-    auto val = ((MAP &)allArgs)[key];
-    if (!val.empty()) {
-        value = (TYPE)val;
-    }
-}
-
-class ProtocolOption {
-public:
-    ProtocolOption();
-
-    enum {
-        kModifyStampOff = 0, // 采用源视频流绝对时间戳，不做任何改变
-        kModifyStampSystem = 1, // 采用zlmediakit接收数据时的系统时间戳(有平滑处理)
-        kModifyStampRelative = 2 // 采用源视频流时间戳相对时间戳(增长量)，有做时间戳跳跃和回退矫正
-    };
-    // 时间戳类型
-    int modify_stamp;
-
-    //转协议是否开启音频
-    bool enable_audio;
-    //添加静音音频，在关闭音频时，此开关无效
-    bool add_mute_audio;
-    // 无人观看时，是否直接关闭(而不是通过on_none_reader hook返回close)
-    // 此配置置1时，此流如果无人观看，将不触发on_none_reader hook回调，
-    // 而是将直接关闭流
-    bool auto_close;
-
-    //断连续推延时，单位毫秒，默认采用配置文件
-    uint32_t continue_push_ms;
-
-    // 平滑发送定时器间隔，单位毫秒，置0则关闭；开启后影响cpu性能同时增加内存
-    // 该配置开启后可以解决一些流发送不平滑导致zlmediakit转发也不平滑的问题
-    uint32_t paced_sender_ms;
-
-    //是否开启转换为hls(mpegts)
-    bool enable_hls;
-    //是否开启转换为hls(fmp4)
-    bool enable_hls_fmp4;
-    //是否开启MP4录制
-    bool enable_mp4;
-    //是否开启转换为rtsp/webrtc
-    bool enable_rtsp;
-    //是否开启转换为rtmp/flv
-    bool enable_rtmp;
-    //是否开启转换为http-ts/ws-ts
-    bool enable_ts;
-    //是否开启转换为http-fmp4/ws-fmp4
-    bool enable_fmp4;
-
-    // hls协议是否按需生成，如果hls.segNum配置为0(意味着hls录制)，那么hls将一直生成(不管此开关)
-    bool hls_demand;
-    // rtsp[s]协议是否按需生成
-    bool rtsp_demand;
-    // rtmp[s]、http[s]-flv、ws[s]-flv协议是否按需生成
-    bool rtmp_demand;
-    // http[s]-ts协议是否按需生成
-    bool ts_demand;
-    // http[s]-fmp4、ws[s]-fmp4协议是否按需生成
-    bool fmp4_demand;
-
-    //是否将mp4录制当做观看者
-    bool mp4_as_player;
-    //mp4切片大小，单位秒
-    size_t mp4_max_second;
-    //mp4录制保存路径
-    std::string mp4_save_path;
-
-    //hls录制保存路径
-    std::string hls_save_path;
-
-    // 支持通过on_publish返回值替换stream_id
-    std::string stream_replace;
-
-    // 最大track数
-    size_t max_track = 2;
-
-    template <typename MAP>
-    ProtocolOption(const MAP &allArgs) : ProtocolOption() {
-        load(allArgs);
-    }
-
-    template <typename MAP>
-    void load(const MAP &allArgs) {
-#define GET_OPT_VALUE(key) getArgsValue(allArgs, #key, key)
-        GET_OPT_VALUE(modify_stamp);
-        GET_OPT_VALUE(enable_audio);
-        GET_OPT_VALUE(add_mute_audio);
-        GET_OPT_VALUE(auto_close);
-        GET_OPT_VALUE(continue_push_ms);
-        GET_OPT_VALUE(paced_sender_ms);
-
-        GET_OPT_VALUE(enable_hls);
-        GET_OPT_VALUE(enable_hls_fmp4);
-        GET_OPT_VALUE(enable_mp4);
-        GET_OPT_VALUE(enable_rtsp);
-        GET_OPT_VALUE(enable_rtmp);
-        GET_OPT_VALUE(enable_ts);
-        GET_OPT_VALUE(enable_fmp4);
-
-        GET_OPT_VALUE(hls_demand);
-        GET_OPT_VALUE(rtsp_demand);
-        GET_OPT_VALUE(rtmp_demand);
-        GET_OPT_VALUE(ts_demand);
-        GET_OPT_VALUE(fmp4_demand);
-
-        GET_OPT_VALUE(mp4_max_second);
-        GET_OPT_VALUE(mp4_as_player);
-        GET_OPT_VALUE(mp4_save_path);
-
-        GET_OPT_VALUE(hls_save_path);
-        GET_OPT_VALUE(stream_replace);
-        GET_OPT_VALUE(max_track);
-    }
-};
-
-//该对象用于拦截感兴趣的MediaSourceEvent事件
-class MediaSourceEventInterceptor : public MediaSourceEvent {
-public:
-    void setDelegate(const std::weak_ptr<MediaSourceEvent> &listener);
-    std::shared_ptr<MediaSourceEvent> getDelegate() const;
-
-    MediaOriginType getOriginType(MediaSource &sender) const override;
-    std::string getOriginUrl(MediaSource &sender) const override;
-    std::shared_ptr<toolkit::SockInfo> getOriginSock(MediaSource &sender) const override;
-
-    bool seekTo(MediaSource &sender, uint32_t stamp) override;
-    bool pause(MediaSource &sender,  bool pause) override;
-    bool speed(MediaSource &sender, float speed) override;
-    bool close(MediaSource &sender) override;
-    int totalReaderCount(MediaSource &sender) override;
-    void onReaderChanged(MediaSource &sender, int size) override;
-    void onRegist(MediaSource &sender, bool regist) override;
-    bool setupRecord(MediaSource &sender, Recorder::type type, bool start, const std::string &custom_path, size_t max_second) override;
-    bool isRecording(MediaSource &sender, Recorder::type type) override;
-    std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const override;
-    void startSendRtp(MediaSource &sender, const SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) override;
-    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;
-
-private:
-    std::weak_ptr<MediaSourceEvent> _listener;
-};
-
-/**
- * 解析url获取媒体相关信息
- */
-class MediaInfo: public MediaTuple {
-public:
-    MediaInfo() = default;
-    MediaInfo(const std::string &url) { parse(url); }
-    void parse(const std::string &url);
-    std::string getUrl() const { return schema + "://" + shortUrl(); }
-
-public:
-    uint16_t port = 0;
-    std::string full_url;
-    std::string schema;
-    std::string host;
-};
-
-bool equalMediaTuple(const MediaTuple& a, const MediaTuple& b);
-
-/**
- * 媒体源，任何rtsp/rtmp的直播流都源自该对象
- */
-class MediaSource: public TrackSource, public std::enable_shared_from_this<MediaSource> {
-public:
-    static MediaSource& NullMediaSource();
-    using Ptr = std::shared_ptr<MediaSource>;
-
-    MediaSource(const std::string &schema, const MediaTuple& tuple);
-    virtual ~MediaSource();
-
-    ////////////////获取MediaSource相关信息////////////////
-
-    // 获取协议类型
-    const std::string& getSchema() const {
-        return _schema;
-    }
-
-    const MediaTuple& getMediaTuple() const {
-        return _tuple;
-    }
-
-    std::string getUrl() const { return _schema + "://" + _tuple.shortUrl(); }
-
-    //获取对象所有权
-    std::shared_ptr<void> getOwnership();
-
-    // 获取所有Track
-    std::vector<Track::Ptr> getTracks(bool ready = true) const override;
-
-    // 获取流当前时间戳
-    virtual uint32_t getTimeStamp(TrackType type) { return 0; };
-    // 设置时间戳
-    virtual void setTimeStamp(uint32_t stamp) {};
-
-    // 获取数据速率，单位bytes/s
-    int getBytesSpeed(TrackType type = TrackInvalid);
-    // 获取流创建GMT unix时间戳，单位秒
-    uint64_t getCreateStamp() const { return _create_stamp; }
-    // 获取流上线时间，单位秒
-    uint64_t getAliveSecond() const;
-
-    ////////////////MediaSourceEvent相关接口实现////////////////
-
-    // 设置监听者
-    virtual void setListener(const std::weak_ptr<MediaSourceEvent> &listener);
-    // 获取监听者
-    std::weak_ptr<MediaSourceEvent> getListener() const;
-
-    // 本协议获取观看者个数，可能返回本协议的观看人数，也可能返回总人数
-    virtual int readerCount() = 0;
-    // 观看者个数，包括(hls/rtsp/rtmp)
-    virtual int totalReaderCount();
-    // 获取播放器列表
-    virtual void getPlayerList(const std::function<void(const std::list<toolkit::Any> &info_list)> &cb,
-                               const std::function<toolkit::Any(toolkit::Any &&info)> &on_change) {
-        assert(cb);
-        cb(std::list<toolkit::Any>());
-    }
-
-    virtual bool broadcastMessage(const toolkit::Any &data) { return false; }
-
-    // 获取媒体源类型
-    MediaOriginType getOriginType() const;
-    // 获取媒体源url或者文件路径
-    std::string getOriginUrl() const;
-    // 获取媒体源客户端相关信息
-    std::shared_ptr<toolkit::SockInfo> getOriginSock() const;
-
-    // 拖动进度条
-    bool seekTo(uint32_t stamp);
-    // 暂停
-    bool pause(bool pause);
-    // 倍数播放
-    bool speed(float speed);
-    // 关闭该流
-    bool close(bool force);
-    // 该流观看人数变化
-    void onReaderChanged(int size);
-    // 开启或关闭录制
-    bool setupRecord(Recorder::type type, bool start, const std::string &custom_path, size_t max_second);
-    // 获取录制状态
-    bool isRecording(Recorder::type type);
-    // 开始发送ps-rtp
-    void startSendRtp(const MediaSourceEvent::SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb);
-    // 停止发送ps-rtp
-    bool stopSendRtp(const std::string &ssrc);
-    // 获取丢包率
-    float getLossRate(mediakit::TrackType type);
-    // 获取所在线程
-    toolkit::EventPoller::Ptr getOwnerPoller();
-    // 获取MultiMediaSourceMuxer对象
-    std::shared_ptr<MultiMediaSourceMuxer> getMuxer();
-
-    ////////////////static方法，查找或生成MediaSource////////////////
-
-    // 同步查找流
-    static Ptr find(const std::string &schema, const std::string &vhost, const std::string &app, const std::string &id, bool from_mp4 = false);
-    static Ptr find(const MediaInfo &info, bool from_mp4 = false) {
-        return find(info.schema, info.vhost, info.app, info.stream, from_mp4);
-    }
-
-    // 忽略schema，同步查找流，可能返回rtmp/rtsp/hls类型
-    static Ptr find(const std::string &vhost, const std::string &app, const std::string &stream_id, bool from_mp4 = false);
-
-    // 异步查找流
-    static void findAsync(const MediaInfo &info, const std::shared_ptr<toolkit::Session> &session, const std::function<void(const Ptr &src)> &cb);
-    // 遍历所有流
-    static void for_each_media(const std::function<void(const Ptr &src)> &cb, const std::string &schema = "", const std::string &vhost = "", const std::string &app = "", const std::string &stream = "");
-    // 从mp4文件生成MediaSource
-    static MediaSource::Ptr createFromMP4(const std::string &schema, const std::string &vhost, const std::string &app, const std::string &stream, const std::string &file_path = "", bool check_app = true);
-
-protected:
-    //媒体注册
-    void regist();
-
-private:
-    // 媒体注销
-    bool unregist();
-    // 触发媒体事件
-    void emitEvent(bool regist);
-
-protected:
-    toolkit::BytesSpeed _speed[TrackMax];
-    MediaTuple _tuple;
-
-private:
-    std::atomic_flag _owned { false };
-    time_t _create_stamp;
-    toolkit::Ticker _ticker;
-    std::string _schema;
-    std::weak_ptr<MediaSourceEvent> _listener;
-    // 对象个数统计
-    toolkit::ObjectStatistic<MediaSource> _statistic;
-};
-
-} /* namespace mediakit */
-#endif //ZLMEDIAKIT_MEDIASOURCE_H

webrtc_player/android/zlm/src/main/cpp/include/Common/MultiMediaSourceMuxer.h

@@ -1,188 +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_MULTIMEDIASOURCEMUXER_H
-#define ZLMEDIAKIT_MULTIMEDIASOURCEMUXER_H
-
-#include "Common/Stamp.h"
-#include "Common/MediaSource.h"
-#include "Common/MediaSink.h"
-#include "Record/Recorder.h"
-#include "Rtp/RtpSender.h"
-#include "Record/HlsRecorder.h"
-#include "Record/HlsMediaSource.h"
-#include "Rtsp/RtspMediaSourceMuxer.h"
-#include "Rtmp/RtmpMediaSourceMuxer.h"
-#include "TS/TSMediaSourceMuxer.h"
-#include "FMP4/FMP4MediaSourceMuxer.h"
-
-namespace mediakit {
-
-class MultiMediaSourceMuxer : public MediaSourceEventInterceptor, public MediaSink, public std::enable_shared_from_this<MultiMediaSourceMuxer>{
-public:
-    using Ptr = std::shared_ptr<MultiMediaSourceMuxer>;
-    using RingType = toolkit::RingBuffer<Frame::Ptr>;
-
-    class Listener {
-    public:
-        virtual ~Listener() = default;
-        virtual void onAllTrackReady() = 0;
-    };
-
-    MultiMediaSourceMuxer(const MediaTuple& tuple, float dur_sec = 0.0,const ProtocolOption &option = ProtocolOption());
-
-    /**
-     * 设置事件监听器
-     * @param listener 监听器
-     */
-    void setMediaListener(const std::weak_ptr<MediaSourceEvent> &listener);
-
-     /**
-      * 设置Track就绪事件监听器
-      * @param listener 事件监听器
-     */
-    void setTrackListener(const std::weak_ptr<Listener> &listener);
-
-    /**
-     * 返回总的消费者个数
-     */
-    int totalReaderCount() const;
-
-    /**
-     * 判断是否生效(是否正在转其他协议)
-     */
-    bool isEnabled();
-
-    /**
-     * 设置MediaSource时间戳
-     * @param stamp 时间戳
-     */
-    void setTimeStamp(uint32_t stamp);
-
-    /**
-     * 重置track
-     */
-    void resetTracks() override;
-
-    /////////////////////////////////MediaSourceEvent override/////////////////////////////////
-
-    /**
-     * 观看总人数
-     * @param sender 事件发送者
-     * @return 观看总人数
-     */
-    int totalReaderCount(MediaSource &sender) override;
-
-    /**
-     * 设置录制状态
-     * @param type 录制类型
-     * @param start 开始或停止
-     * @param custom_path 开启录制时，指定自定义路径
-     * @return 是否设置成功
-     */
-    bool setupRecord(MediaSource &sender, Recorder::type type, bool start, const std::string &custom_path, size_t max_second) override;
-
-    /**
-     * 获取录制状态
-     * @param type 录制类型
-     * @return 录制状态
-     */
-    bool isRecording(MediaSource &sender, Recorder::type type) override;
-
-    /**
-     * 开始发送ps-rtp流
-     * @param dst_url 目标ip或域名
-     * @param dst_port 目标端口
-     * @param ssrc rtp的ssrc
-     * @param is_udp 是否为udp
-     * @param cb 启动成功或失败回调
-     */
-    void startSendRtp(MediaSource &sender, const MediaSourceEvent::SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) override;
-
-    /**
-     * 停止ps-rtp发送
-     * @return 是否成功
-     */
-    bool stopSendRtp(MediaSource &sender, const std::string &ssrc) override;
-
-    /**
-     * 获取所有Track
-     * @param trackReady 是否筛选过滤未就绪的track
-     * @return 所有Track
-     */
-    std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const override;
-
-    /**
-     * 获取所属线程
-     */
-    toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override;
-
-    /**
-     * 获取本对象
-     */
-    std::shared_ptr<MultiMediaSourceMuxer> getMuxer(MediaSource &sender) override;
-
-    const ProtocolOption &getOption() const;
-    const MediaTuple &getMediaTuple() const;
-    std::string shortUrl() const;
-
-protected:
-    /////////////////////////////////MediaSink override/////////////////////////////////
-
-    /**
-    * 某track已经准备好，其ready()状态返回true，
-    * 此时代表可以获取其例如sps pps等相关信息了
-    * @param track
-    */
-    bool onTrackReady(const Track::Ptr & track) override;
-
-    /**
-     * 所有Track已经准备好，
-     */
-    void onAllTrackReady() override;
-
-    /**
-     * 某Track输出frame，在onAllTrackReady触发后才会调用此方法
-     * @param frame
-     */
-    bool onTrackFrame(const Frame::Ptr &frame) override;
-    bool onTrackFrame_l(const Frame::Ptr &frame);
-
-private:
-    void createGopCacheIfNeed();
-
-private:
-    bool _is_enable = false;
-    bool _create_in_poller = false;
-    bool _video_key_pos = false;
-    float _dur_sec;
-    std::shared_ptr<class FramePacedSender> _paced_sender;
-    MediaTuple _tuple;
-    ProtocolOption _option;
-    toolkit::Ticker _last_check;
-    std::unordered_map<int, Stamp> _stamps;
-    std::weak_ptr<Listener> _track_listener;
-    std::unordered_multimap<std::string, RingType::RingReader::Ptr> _rtp_sender;
-    FMP4MediaSourceMuxer::Ptr _fmp4;
-    RtmpMediaSourceMuxer::Ptr _rtmp;
-    RtspMediaSourceMuxer::Ptr _rtsp;
-    TSMediaSourceMuxer::Ptr _ts;
-    MediaSinkInterface::Ptr _mp4;
-    HlsRecorder::Ptr _hls;
-    HlsFMP4Recorder::Ptr _hls_fmp4;
-    toolkit::EventPoller::Ptr _poller;
-    RingType::Ptr _ring;
-
-    //对象个数统计
-    toolkit::ObjectStatistic<MultiMediaSourceMuxer> _statistic;
-};
-
-}//namespace mediakit
-#endif //ZLMEDIAKIT_MULTIMEDIASOURCEMUXER_H

webrtc_player/android/zlm/src/main/cpp/include/Common/PacketCache.h

@@ -1,90 +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_PACKET_CACHE_H_
-#define ZLMEDIAKIT_PACKET_CACHE_H_
-
-#include "Common/config.h"
-#include "Util/List.h"
-
-namespace mediakit {
-/// 缓存刷新策略类
-class FlushPolicy {
-public:
-    bool isFlushAble(bool is_video, bool is_key, uint64_t new_stamp, size_t cache_size);
-
-private:
-    // 音视频的最后时间戳
-    uint64_t _last_stamp[2] = { 0, 0 };
-};
-
-/// 合并写缓存模板
-/// \tparam packet 包类型
-/// \tparam policy 刷新缓存策略
-/// \tparam packet_list 包缓存类型
-template<typename packet, typename policy = FlushPolicy, typename packet_list = toolkit::List<std::shared_ptr<packet> > >
-class PacketCache {
-public:
-    PacketCache() { _cache = std::make_shared<packet_list>(); }
-
-    virtual ~PacketCache() = default;
-
-    void inputPacket(uint64_t stamp, bool is_video, std::shared_ptr<packet> pkt, bool key_pos) {
-        bool flag = flushImmediatelyWhenCloseMerge();
-        if (!flag && _policy.isFlushAble(is_video, key_pos, stamp, _cache->size())) {
-            flush();
-        }
-
-        //追加数据到最后
-        _cache->emplace_back(std::move(pkt));
-        if (key_pos) {
-            _key_pos = key_pos;
-        }
-
-        if (flag) {
-            flush();
-        }
-    }
-
-    void flush() {
-        if (_cache->empty()) {
-            return;
-        }
-        onFlush(std::move(_cache), _key_pos);
-        _cache = std::make_shared<packet_list>();
-        _key_pos = false;
-    }
-
-    virtual void clearCache() {
-        _cache->clear();
-    }
-
-    virtual void onFlush(std::shared_ptr<packet_list>, bool key_pos) = 0;
-
-private:
-    bool flushImmediatelyWhenCloseMerge() {
-        // 一般的协议关闭合并写时，立即刷新缓存，这样可以减少一帧的延时，但是rtp例外
-        // 因为rtp的包很小，一个RtpPacket包中也不是完整的一帧图像，所以在关闭合并写时，
-        // 还是有必要缓冲一帧的rtp(也就是时间戳相同的rtp)再输出，这样虽然会增加一帧的延时
-        // 但是却对性能提升很大，这样做还是比较划算的
-
-        GET_CONFIG(int, mergeWriteMS, General::kMergeWriteMS);
-        GET_CONFIG(int, rtspLowLatency, Rtsp::kLowLatency);
-        return std::is_same<packet, RtpPacket>::value ? rtspLowLatency : (mergeWriteMS <= 0);
-    }
-
-private:
-    bool _key_pos = false;
-    policy _policy;
-    std::shared_ptr<packet_list> _cache;
-};
-}
-
-#endif //ZLMEDIAKIT_PACKET_CACHE_H_

webrtc_player/android/zlm/src/main/cpp/include/Common/Parser.h

@@ -1,138 +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_PARSER_H
-#define ZLMEDIAKIT_PARSER_H
-
-#include <map>
-#include <string>
-#include "Util/util.h"
-
-namespace mediakit {
-
-// 从字符串中提取子字符串
-std::string findSubString(const char *buf, const char *start, const char *end, size_t buf_size = 0);
-// 把url解析为主机地址和端口号,兼容ipv4/ipv6/dns
-void splitUrl(const std::string &url, std::string &host, uint16_t &port);
-// 解析proxy url,仅支持http
-void parseProxyUrl(const std::string &proxy_url, std::string &proxy_host, uint16_t &proxy_port, std::string &proxy_auth);
-
-struct StrCaseCompare {
-    bool operator()(const std::string &__x, const std::string &__y) const { return strcasecmp(__x.data(), __y.data()) < 0; }
-};
-
-class StrCaseMap : public std::multimap<std::string, std::string, StrCaseCompare> {
-public:
-    using Super = std::multimap<std::string, std::string, StrCaseCompare>;
-
-    std::string &operator[](const std::string &k) {
-        auto it = find(k);
-        if (it == end()) {
-            it = Super::emplace(k, "");
-        }
-        return it->second;
-    }
-
-    template <typename K, typename V>
-    void emplace(K &&k, V &&v) {
-        auto it = find(k);
-        if (it != end()) {
-            return;
-        }
-        Super::emplace(std::forward<K>(k), std::forward<V>(v));
-    }
-
-    template <typename K, typename V>
-    void emplace_force(K &&k, V &&v) {
-        Super::emplace(std::forward<K>(k), std::forward<V>(v));
-    }
-};
-
-// rtsp/http/sip解析类
-class Parser {
-public:
-    // 解析http/rtsp/sip请求，需要确保buf以\0结尾
-    void parse(const char *buf, size_t size);
-
-    // 获取命令字，如GET/POST
-    const std::string &method() const;
-
-    // 请求时，获取中间url，不包含?后面的参数
-    const std::string &url() const;
-    // 回复时，获取状态码，如200/404
-    const std::string &status() const;
-
-    // 获取中间url，包含?后面的参数
-    std::string fullUrl() const;
-
-    // 请求时，获取协议名，如HTTP/1.1
-    const std::string &protocol() const;
-    // 回复时，获取状态字符串，如 OK/Not Found
-    const std::string &statusStr() const;
-
-    // 根据header key名，获取请求header value值
-    const std::string &operator[](const char *name) const;
-
-    // 获取http body或sdp
-    const std::string &content() const;
-
-    // 清空，为了重用
-    void clear();
-
-    // 获取?后面的参数
-    const std::string &params() const;
-
-    // 重新设置url
-    void setUrl(std::string url);
-
-    // 重新设置content
-    void setContent(std::string content);
-
-    // 获取header列表
-    StrCaseMap &getHeader() const;
-
-    // 获取url参数列表
-    StrCaseMap &getUrlArgs() const;
-
-    // 解析?后面的参数
-    static StrCaseMap parseArgs(const std::string &str, const char *pair_delim = "&", const char *key_delim = "=");
-
-    static std::string mergeUrl(const std::string &base_url, const std::string &path);
-
-private:
-    std::string _method;
-    std::string _url;
-    std::string _protocol;
-    std::string _content;
-    std::string _params;
-    mutable StrCaseMap _headers;
-    mutable StrCaseMap _url_args;
-};
-
-// 解析rtsp url的工具类
-class RtspUrl {
-public:
-    bool _is_ssl;
-    uint16_t _port;
-    std::string _url;
-    std::string _user;
-    std::string _passwd;
-    std::string _host;
-
-public:
-    void parse(const std::string &url);
-
-private:
-    void setup(bool, const std::string &, const std::string &, const std::string &);
-};
-
-} // namespace mediakit
-
-#endif // ZLMEDIAKIT_PARSER_H

webrtc_player/android/zlm/src/main/cpp/include/Common/Stamp.h

@@ -1,148 +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_STAMP_H
-#define ZLMEDIAKIT_STAMP_H
-
-#include <set>
-#include <cstdint>
-#include "Util/TimeTicker.h"
-
-namespace mediakit {
-
-class DeltaStamp {
-public:
-    DeltaStamp();
-    virtual ~DeltaStamp() = default;
-
-    /**
-     * 计算时间戳增量
-     * @param stamp 绝对时间戳
-     * @param enable_rollback 是否允许相当时间戳回退
-     * @return 时间戳增量
-     */
-    int64_t deltaStamp(int64_t stamp, bool enable_rollback = true);
-    int64_t relativeStamp(int64_t stamp, bool enable_rollback = true);
-    int64_t relativeStamp();
-
-    // 设置最大允许回退或跳跃幅度
-    void setMaxDelta(size_t max_delta);
-
-protected:
-    virtual void needSync() {}
-
-protected:
-    int _max_delta;
-    int64_t _last_stamp = 0;
-    int64_t _relative_stamp = 0;
-};
-
-//该类解决时间戳回环、回退问题
-//计算相对时间戳或者产生平滑时间戳
-class Stamp : public DeltaStamp{
-public:
-    /**
-     * 求取相对时间戳,同时实现了音视频同步、限制dts回退等功能
-     * @param dts 输入dts，如果为0则根据系统时间戳生成
-     * @param pts 输入pts，如果为0则等于dts
-     * @param dts_out 输出dts
-     * @param pts_out 输出pts
-     * @param modifyStamp 是否用系统时间戳覆盖
-     */
-    void revise(int64_t dts, int64_t pts, int64_t &dts_out, int64_t &pts_out,bool modifyStamp = false);
-
-    /**
-     * 再设置相对时间戳，用于seek用
-     * @param relativeStamp 相对时间戳
-     */
-    void setRelativeStamp(int64_t relativeStamp);
-
-    /**
-     * 获取当前相对时间戳
-     * @return
-     */
-    int64_t getRelativeStamp() const ;
-
-    /**
-     * 设置是否为回放模式，回放模式运行时间戳回退
-     * @param playback 是否为回放模式
-     */
-    void setPlayBack(bool playback = true);
-
-    /**
-     * 音视频同步用，音频应该同步于视频(只修改音频时间戳)
-     * 因为音频时间戳修改后不影响播放速度
-     */
-    void syncTo(Stamp &other);
-
-    /**
-     * 是否允许时间戳回退
-     */
-    void enableRollback(bool flag);
-
-private:
-    //主要实现音视频时间戳同步功能
-    void revise_l(int64_t dts, int64_t pts, int64_t &dts_out, int64_t &pts_out,bool modifyStamp = false);
-
-    //主要实现获取相对时间戳功能
-    void revise_l2(int64_t dts, int64_t pts, int64_t &dts_out, int64_t &pts_out,bool modifyStamp = false);
-
-    void needSync() override;
-
-private:
-    bool _playback = false;
-    bool _need_sync = false;
-    // 默认不允许时间戳回滚
-    bool _enable_rollback = false;
-    int64_t _relative_stamp = 0;
-    int64_t _last_dts_in = 0;
-    int64_t _last_dts_out = 0;
-    int64_t _last_pts_out = 0;
-    toolkit::SmoothTicker _ticker;
-    Stamp *_sync_master = nullptr;
-};
-
-//dts生成器，
-//pts排序后就是dts
-class DtsGenerator{
-public:
-    bool getDts(uint64_t pts, uint64_t &dts);
-
-private:
-    bool getDts_l(uint64_t pts, uint64_t &dts);
-
-private:
-    uint64_t _dts_pts_offset = 0;
-    uint64_t _last_dts = 0;
-    uint64_t _last_pts = 0;
-    uint64_t _last_max_pts = 0;
-    size_t _frames_since_last_max_pts = 0;
-    size_t _sorter_max_size = 0;
-    size_t _count_sorter_max_size = 0;
-    std::set<uint64_t> _pts_sorter;
-};
-
-class NtpStamp {
-public:
-    void setNtpStamp(uint32_t rtp_stamp, uint64_t ntp_stamp_ms);
-    uint64_t getNtpStamp(uint32_t rtp_stamp, uint32_t sample_rate);
-
-private:
-    void update(uint32_t rtp_stamp, uint64_t ntp_stamp_us);
-    uint64_t getNtpStampUS(uint32_t rtp_stamp, uint32_t sample_rate);
-
-private:
-    uint32_t _last_rtp_stamp = 0;
-    uint64_t _last_ntp_stamp_us = 0;
-};
-
-}//namespace mediakit
-
-#endif //ZLMEDIAKIT_STAMP_H

webrtc_player/android/zlm/src/main/cpp/include/Common/config.h

@@ -1,444 +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 COMMON_CONFIG_H
-#define COMMON_CONFIG_H
-
-#include "Util/NoticeCenter.h"
-#include "Util/mini.h"
-#include "Util/onceToken.h"
-#include "macros.h"
-#include <functional>
-
-namespace mediakit {
-
-class ProtocolOption;
-
-// 加载配置文件，如果配置文件不存在，那么会导出默认配置并生成配置文件
-// 加载配置文件成功后会触发kBroadcastUpdateConfig广播
-// 如果指定的文件名(ini_path)为空，那么会加载默认配置文件
-// 默认配置文件名为 /path/to/your/exe.ini
-// 加载配置文件成功后返回true，否则返回false
-bool loadIniConfig(const char *ini_path = nullptr);
-
-////////////广播名称///////////
-namespace Broadcast {
-
-// 注册或反注册MediaSource事件广播
-extern const std::string kBroadcastMediaChanged;
-#define BroadcastMediaChangedArgs const bool &bRegist, MediaSource &sender
-
-// 录制mp4文件成功后广播
-extern const std::string kBroadcastRecordMP4;
-#define BroadcastRecordMP4Args const RecordInfo &info
-
-// 录制 ts 文件后广播
-extern const std::string kBroadcastRecordTs;
-#define BroadcastRecordTsArgs const RecordInfo &info
-
-// 收到http api请求广播
-extern const std::string kBroadcastHttpRequest;
-#define BroadcastHttpRequestArgs const Parser &parser, const HttpSession::HttpResponseInvoker &invoker, bool &consumed, SockInfo &sender
-
-// 在http文件服务器中,收到http访问文件或目录的广播,通过该事件控制访问http目录的权限
-extern const std::string kBroadcastHttpAccess;
-#define BroadcastHttpAccessArgs const Parser &parser, const std::string &path, const bool &is_dir, const HttpSession::HttpAccessPathInvoker &invoker, SockInfo &sender
-
-// 在http文件服务器中,收到http访问文件或目录前的广播,通过该事件可以控制http url到文件路径的映射
-// 在该事件中通过自行覆盖path参数，可以做到譬如根据虚拟主机或者app选择不同http根目录的目的
-extern const std::string kBroadcastHttpBeforeAccess;
-#define BroadcastHttpBeforeAccessArgs const Parser &parser, std::string &path, SockInfo &sender
-
-// 该流是否需要认证？是的话调用invoker并传入realm,否则传入空的realm.如果该事件不监听则不认证
-extern const std::string kBroadcastOnGetRtspRealm;
-#define BroadcastOnGetRtspRealmArgs const MediaInfo &args, const RtspSession::onGetRealm &invoker, SockInfo &sender
-
-// 请求认证用户密码事件，user_name为用户名，must_no_encrypt如果为true，则必须提供明文密码(因为此时是base64认证方式),否则会导致认证失败
-// 获取到密码后请调用invoker并输入对应类型的密码和密码类型，invoker执行时会匹配密码
-extern const std::string kBroadcastOnRtspAuth;
-#define BroadcastOnRtspAuthArgs const MediaInfo &args, const std::string &realm, const std::string &user_name, const bool &must_no_encrypt, const RtspSession::onAuth &invoker, SockInfo &sender
-
-// 推流鉴权结果回调对象
-// 如果err为空则代表鉴权成功
-using PublishAuthInvoker = std::function<void(const std::string &err, const ProtocolOption &option)>;
-
-// 收到rtsp/rtmp推流事件广播，通过该事件控制推流鉴权
-extern const std::string kBroadcastMediaPublish;
-#define BroadcastMediaPublishArgs const MediaOriginType &type, const MediaInfo &args, const Broadcast::PublishAuthInvoker &invoker, SockInfo &sender
-
-// 播放鉴权结果回调对象
-// 如果err为空则代表鉴权成功
-using AuthInvoker = std::function<void(const std::string &err)>;
-
-// 播放rtsp/rtmp/http-flv事件广播，通过该事件控制播放鉴权
-extern const std::string kBroadcastMediaPlayed;
-#define BroadcastMediaPlayedArgs const MediaInfo &args, const Broadcast::AuthInvoker &invoker, SockInfo &sender
-
-// shell登录鉴权
-extern const std::string kBroadcastShellLogin;
-#define BroadcastShellLoginArgs const std::string &user_name, const std::string &passwd, const Broadcast::AuthInvoker &invoker, SockInfo &sender
-
-// 停止rtsp/rtmp/http-flv会话后流量汇报事件广播
-extern const std::string kBroadcastFlowReport;
-#define BroadcastFlowReportArgs const MediaInfo &args, const uint64_t &totalBytes, const uint64_t &totalDuration, const bool &isPlayer, SockInfo &sender
-
-// 未找到流后会广播该事件，请在监听该事件后去拉流或其他方式产生流，这样就能按需拉流了
-extern const std::string kBroadcastNotFoundStream;
-#define BroadcastNotFoundStreamArgs const MediaInfo &args, SockInfo &sender, const std::function<void()> &closePlayer
-
-// 某个流无人消费时触发，目的为了实现无人观看时主动断开拉流等业务逻辑
-extern const std::string kBroadcastStreamNoneReader;
-#define BroadcastStreamNoneReaderArgs MediaSource &sender
-
-// rtp推流被动停止时触发
-extern const std::string kBroadcastSendRtpStopped;
-#define BroadcastSendRtpStoppedArgs MultiMediaSourceMuxer &sender, const std::string &ssrc, const SockException &ex
-
-// 更新配置文件事件广播,执行loadIniConfig函数加载配置文件成功后会触发该广播
-extern const std::string kBroadcastReloadConfig;
-#define BroadcastReloadConfigArgs void
-
-// rtp server 超时
-extern const std::string kBroadcastRtpServerTimeout;
-#define BroadcastRtpServerTimeoutArgs uint16_t &local_port, const string &stream_id,int &tcp_mode, bool &re_use_port, uint32_t &ssrc
-
-// rtc transport sctp 连接状态
-extern const std::string kBroadcastRtcSctpConnecting;
-extern const std::string kBroadcastRtcSctpConnected;
-extern const std::string kBroadcastRtcSctpFailed;
-extern const std::string kBroadcastRtcSctpClosed;
-#define BroadcastRtcSctpConnectArgs WebRtcTransport& sender
-
-// rtc transport sctp 发送数据
-extern const std::string kBroadcastRtcSctpSend;
-#define BroadcastRtcSctpSendArgs WebRtcTransport& sender, const uint8_t *&data, size_t& len
-
-// rtc transport sctp 接收数据
-extern const std::string kBroadcastRtcSctpReceived;
-#define BroadcastRtcSctpReceivedArgs WebRtcTransport& sender, uint16_t &streamId, uint32_t &ppid, const uint8_t *&msg, size_t &len
-
-#define ReloadConfigTag ((void *)(0xFF))
-#define RELOAD_KEY(arg, key)                                                                                           \
-    do {                                                                                                               \
-        decltype(arg) arg##_tmp = ::toolkit::mINI::Instance()[key];                                                    \
-        if (arg == arg##_tmp) {                                                                                        \
-            return;                                                                                                    \
-        }                                                                                                              \
-        arg = arg##_tmp;                                                                                               \
-        InfoL << "reload config:" << key << "=" << arg;                                                                \
-    } while (0)
-
-// 监听某个配置发送变更
-#define LISTEN_RELOAD_KEY(arg, key, ...)                                                                               \
-    do {                                                                                                               \
-        static ::toolkit::onceToken s_token_listen([]() {                                                              \
-            ::toolkit::NoticeCenter::Instance().addListener(                                                           \
-                ReloadConfigTag, Broadcast::kBroadcastReloadConfig, [](BroadcastReloadConfigArgs) { __VA_ARGS__; });   \
-        });                                                                                                            \
-    } while (0)
-
-#define GET_CONFIG(type, arg, key)                                                                                     \
-    static type arg = ::toolkit::mINI::Instance()[key];                                                                \
-    LISTEN_RELOAD_KEY(arg, key, { RELOAD_KEY(arg, key); });
-
-#define GET_CONFIG_FUNC(type, arg, key, ...)                                                                           \
-    static type arg;                                                                                                   \
-    do {                                                                                                               \
-        static ::toolkit::onceToken s_token_set([]() {                                                                 \
-            static auto lam = __VA_ARGS__;                                                                             \
-            static auto arg##_str = ::toolkit::mINI::Instance()[key];                                                  \
-            arg = lam(arg##_str);                                                                                      \
-            LISTEN_RELOAD_KEY(arg, key, {                                                                              \
-                RELOAD_KEY(arg##_str, key);                                                                            \
-                arg = lam(arg##_str);                                                                                  \
-            });                                                                                                        \
-        });                                                                                                            \
-    } while (0)
-
-} // namespace Broadcast
-
-////////////通用配置///////////
-namespace General {
-// 每个流媒体服务器的ID（GUID）
-extern const std::string kMediaServerId;
-// 流量汇报事件流量阈值,单位KB，默认1MB
-extern const std::string kFlowThreshold;
-// 流无人观看并且超过若干时间后才触发kBroadcastStreamNoneReader事件
-// 默认连续5秒无人观看然后触发kBroadcastStreamNoneReader事件
-extern const std::string kStreamNoneReaderDelayMS;
-// 等待流注册超时时间，收到播放器后请求后，如果未找到相关流，服务器会等待一定时间，
-// 如果在这个时间内，相关流注册上了，那么服务器会立即响应播放器播放成功，
-// 否则会最多等待kMaxStreamWaitTimeMS毫秒，然后响应播放器播放失败
-extern const std::string kMaxStreamWaitTimeMS;
-// 是否启动虚拟主机
-extern const std::string kEnableVhost;
-// 拉流代理时如果断流再重连成功是否删除前一次的媒体流数据，如果删除将重新开始，
-// 如果不删除将会接着上一次的数据继续写(录制hls/mp4时会继续在前一个文件后面写)
-extern const std::string kResetWhenRePlay;
-// 合并写缓存大小(单位毫秒)，合并写指服务器缓存一定的数据后才会一次性写入socket，这样能提高性能，但是会提高延时
-// 开启后会同时关闭TCP_NODELAY并开启MSG_MORE
-extern const std::string kMergeWriteMS;
-// 在docker环境下，不能通过英伟达驱动是否存在来判断是否支持硬件转码
-extern const std::string kCheckNvidiaDev;
-// 是否开启ffmpeg日志
-extern const std::string kEnableFFmpegLog;
-// 最多等待未初始化的Track 10秒，超时之后会忽略未初始化的Track
-extern const std::string kWaitTrackReadyMS;
-// 如果直播流只有单Track，最多等待3秒，超时后未收到其他Track的数据，则认为是单Track
-// 如果协议元数据有声明特定track数，那么无此等待时间
-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;
-} // namespace General
-
-namespace Protocol {
-//时间戳修复这一路流标志位
-extern const std::string kModifyStamp;
-//转协议是否开启音频
-extern const std::string kEnableAudio;
-//添加静音音频，在关闭音频时，此开关无效
-extern const std::string kAddMuteAudio;
-// 无人观看时，是否直接关闭(而不是通过on_none_reader hook返回close)
-// 此配置置1时，此流如果无人观看，将不触发on_none_reader hook回调，
-// 而是将直接关闭流
-extern const std::string kAutoClose;
-//断连续推延时，单位毫秒，默认采用配置文件
-extern const std::string kContinuePushMS;
-// 平滑发送定时器间隔，单位毫秒，置0则关闭；开启后影响cpu性能同时增加内存
-// 该配置开启后可以解决一些流发送不平滑导致zlmediakit转发也不平滑的问题
-extern const std::string kPacedSenderMS;
-
-//是否开启转换为hls(mpegts)
-extern const std::string kEnableHls;
-//是否开启转换为hls(fmp4)
-extern const std::string kEnableHlsFmp4;
-//是否开启MP4录制
-extern const std::string kEnableMP4;
-//是否开启转换为rtsp/webrtc
-extern const std::string kEnableRtsp;
-//是否开启转换为rtmp/flv
-extern const std::string kEnableRtmp;
-//是否开启转换为http-ts/ws-ts
-extern const std::string kEnableTS;
-//是否开启转换为http-fmp4/ws-fmp4
-extern const std::string kEnableFMP4;
-
-//是否将mp4录制当做观看者
-extern const std::string kMP4AsPlayer;
-//mp4切片大小，单位秒
-extern const std::string kMP4MaxSecond;
-//mp4录制保存路径
-extern const std::string kMP4SavePath;
-
-//hls录制保存路径
-extern const std::string kHlsSavePath;
-
-// 按需转协议的开关
-extern const std::string kHlsDemand;
-extern const std::string kRtspDemand;
-extern const std::string kRtmpDemand;
-extern const std::string kTSDemand;
-extern const std::string kFMP4Demand;
-} // !Protocol
-
-////////////HTTP配置///////////
-namespace Http {
-// http 文件发送缓存大小
-extern const std::string kSendBufSize;
-// http 最大请求字节数
-extern const std::string kMaxReqSize;
-// http keep-alive秒数
-extern const std::string kKeepAliveSecond;
-// http 字符编码
-extern const std::string kCharSet;
-// http 服务器根目录
-extern const std::string kRootPath;
-// http 服务器虚拟目录 虚拟目录名和文件路径使用","隔开，多个配置路径间用";"隔开，例如  path_d,d:/record;path_e,e:/record
-extern const std::string kVirtualPath;
-// http 404错误提示内容
-extern const std::string kNotFound;
-// 是否显示文件夹菜单
-extern const std::string kDirMenu;
-// 禁止缓存文件的后缀
-extern const std::string kForbidCacheSuffix;
-// 可以把http代理前真实客户端ip放在http头中：https://github.com/ZLMediaKit/ZLMediaKit/issues/1388
-extern const std::string kForwardedIpHeader;
-// 是否允许所有跨域请求
-extern const std::string kAllowCrossDomains;
-// 允许访问http api和http文件索引的ip地址范围白名单，置空情况下不做限制
-extern const std::string kAllowIPRange;
-} // namespace Http
-
-////////////SHELL配置///////////
-namespace Shell {
-extern const std::string kMaxReqSize;
-} // namespace Shell
-
-////////////RTSP服务器配置///////////
-namespace Rtsp {
-// 是否优先base64方式认证？默认Md5方式认证
-extern const std::string kAuthBasic;
-// 握手超时时间，默认15秒
-extern const std::string kHandshakeSecond;
-// 维持链接超时时间，默认15秒
-extern const std::string kKeepAliveSecond;
-
-// rtsp拉流代理是否直接代理
-// 直接代理后支持任意编码格式，但是会导致GOP缓存无法定位到I帧，可能会导致开播花屏
-// 并且如果是tcp方式拉流，如果rtp大于mtu会导致无法使用udp方式代理
-// 假定您的拉流源地址不是264或265或AAC，那么你可以使用直接代理的方式来支持rtsp代理
-// 默认开启rtsp直接代理，rtmp由于没有这些问题，是强制开启直接代理的
-extern const std::string kDirectProxy;
-
-// rtsp 转发是否使用低延迟模式，当开启时，不会缓存rtp包，来提高并发，可以降低一帧的延迟
-extern const std::string kLowLatency;
-
-//强制协商rtp传输方式 (0:TCP,1:UDP,2:MULTICAST,-1:不限制)
-//当客户端发起RTSP SETUP的时候如果传输类型和此配置不一致则返回461 Unsupport Transport
-//迫使客户端重新SETUP并切换到对应协议。目前支持FFMPEG和VLC
-extern const std::string kRtpTransportType;
-} // namespace Rtsp
-
-////////////RTMP服务器配置///////////
-namespace Rtmp {
-// 握手超时时间，默认15秒
-extern const std::string kHandshakeSecond;
-// 维持链接超时时间，默认15秒
-extern const std::string kKeepAliveSecond;
-// 是否直接代理
-extern const std::string kDirectProxy;
-// h265-rtmp是否采用增强型(或者国内扩展)
-extern const std::string kEnhanced;
-} // namespace Rtmp
-
-////////////RTP配置///////////
-namespace Rtp {
-// RTP打包最大MTU,公网情况下更小
-extern const std::string kVideoMtuSize;
-// RTP打包最大MTU,公网情况下更小
-extern const std::string kAudioMtuSize;
-// rtp包最大长度限制, 单位KB
-extern const std::string kRtpMaxSize;
-// rtp 打包时，低延迟开关，默认关闭（为0），h264存在一帧多个slice（NAL）的情况，在这种情况下，如果开启可能会导致画面花屏
-extern const std::string kLowLatency;
-//H264 rtp打包模式是否采用stap-a模式(为了在老版本浏览器上兼容webrtc)还是采用Single NAL unit packet per H.264 模式
-extern const std::string kH264StapA;
-} // namespace Rtp
-
-////////////组播配置///////////
-namespace MultiCast {
-// 组播分配起始地址
-extern const std::string kAddrMin;
-// 组播分配截止地址
-extern const std::string kAddrMax;
-// 组播TTL
-extern const std::string kUdpTTL;
-} // namespace MultiCast
-
-////////////录像配置///////////
-namespace Record {
-// 查看录像的应用名称
-extern const std::string kAppName;
-// 每次流化MP4文件的时长,单位毫秒
-extern const std::string kSampleMS;
-// mp4文件写缓存大小
-extern const std::string kFileBufSize;
-// mp4录制完成后是否进行二次关键帧索引写入头部
-extern const std::string kFastStart;
-// mp4文件是否重头循环读取
-extern const std::string kFileRepeat;
-// mp4录制文件是否采用fmp4格式
-extern const std::string kEnableFmp4;
-} // namespace Record
-
-////////////HLS相关配置///////////
-namespace Hls {
-// HLS切片时长,单位秒
-extern const std::string kSegmentDuration;
-// m3u8文件中HLS切片个数，如果设置为0，则不删除切片，而是保存为点播
-extern const std::string kSegmentNum;
-// 如果设置为0，则不保留切片，设置为1则一直保留切片
-extern const std::string kSegmentKeep;
-// HLS切片延迟个数，大于0将生成hls_delay.m3u8文件，0则不生成
-extern const std::string kSegmentDelay;
-// HLS切片从m3u8文件中移除后，继续保留在磁盘上的个数
-extern const std::string kSegmentRetain;
-// HLS文件写缓存大小
-extern const std::string kFileBufSize;
-// 是否广播 ts 切片完成通知
-extern const std::string kBroadcastRecordTs;
-// hls直播文件删除延时，单位秒
-extern const std::string kDeleteDelaySec;
-// 如果设置为1，则第一个切片长度强制设置为1个GOP
-extern const std::string kFastRegister;
-} // namespace Hls
-
-////////////Rtp代理相关配置///////////
-namespace RtpProxy {
-// rtp调试数据保存目录,置空则不生成
-extern const std::string kDumpDir;
-// rtp接收超时时间
-extern const std::string kTimeoutSec;
-// 随机端口范围，最少确保36个端口
-// 该范围同时限制rtsp服务器udp端口范围
-extern const std::string kPortRange;
-// rtp server h264的pt
-extern const std::string kH264PT;
-// rtp server h265的pt
-extern const std::string kH265PT;
-// rtp server ps 的pt
-extern const std::string kPSPT;
-// rtp server opus 的pt
-extern const std::string kOpusPT;
-// RtpSender相关功能是否提前开启gop缓存优化级联秒开体验，默认开启
-extern const std::string kGopCache;
-//国标发送g711 rtp 打包时，每个包的语音时长是多少，默认是100 ms，范围为20~180ms (gb28181-2016，c.2.4规定)，
-//最好为20 的倍数，程序自动向20的倍数取整
-extern const std::string kRtpG711DurMs;
-// udp recv socket buffer size
-extern const std::string kUdpRecvSocketBuffer;
-} // namespace RtpProxy
-
-/**
- * rtsp/rtmp播放器、推流器相关设置名，
- * 这些设置项都不是配置文件用
- * 只用于设置某个播放器或推流器实例用
- */
-namespace Client {
-// 指定网卡ip
-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)
-extern const std::string kRtspPwd;
-// rtsp认证用用户密码是否为md5类型
-extern const std::string kRtspPwdIsMD5;
-// 握手超时时间，默认10,000 毫秒
-extern const std::string kTimeoutMS;
-// rtp/rtmp包接收超时时间，默认5000秒
-extern const std::string kMediaTimeoutMS;
-// rtsp/rtmp心跳时间,默认5000毫秒
-extern const std::string kBeatIntervalMS;
-// 是否为性能测试模式，性能测试模式开启后不会解析rtp或rtmp包
-extern const std::string kBenchmarkMode;
-// 播放器在触发播放成功事件时，是否等待所有track ready时再回调
-extern const std::string kWaitTrackReady;
-// rtsp播放指定track，可选项有0(不指定，默认)、1(视频)、2(音频)
-// 设置方法:player[Client::kPlayTrack] = 0/1/2;
-extern const std::string kPlayTrack;
-//设置代理url，目前只支持http协议
-extern const std::string kProxyUrl;
-} // namespace Client
-} // namespace mediakit
-
-#endif /* COMMON_CONFIG_H */

webrtc_player/android/zlm/src/main/cpp/include/Common/macros.h

@@ -1,102 +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_MACROS_H
-#define ZLMEDIAKIT_MACROS_H
-
-#include "Util/logger.h"
-#include <iostream>
-#include <sstream>
-#if defined(__MACH__)
-#include <arpa/inet.h>
-#include <machine/endian.h>
-#define __BYTE_ORDER BYTE_ORDER
-#define __BIG_ENDIAN BIG_ENDIAN
-#define __LITTLE_ENDIAN LITTLE_ENDIAN
-#elif defined(__linux__)
-#include <arpa/inet.h>
-#include <endian.h>
-#elif defined(_WIN32)
-#define BIG_ENDIAN 1
-#define LITTLE_ENDIAN 0
-#define BYTE_ORDER LITTLE_ENDIAN
-#define __BYTE_ORDER BYTE_ORDER
-#define __BIG_ENDIAN BIG_ENDIAN
-#define __LITTLE_ENDIAN LITTLE_ENDIAN
-#endif
-
-#ifndef CHECK
-#define CHECK(exp, ...) ::mediakit::Assert_ThrowCpp(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__, ##__VA_ARGS__)
-#endif // CHECK
-
-#ifndef CHECK_RET
-#define CHECK_RET(...)                                                         \
-    try {                                                                      \
-        CHECK(__VA_ARGS__);                                                    \
-    } catch (AssertFailedException & ex) {                                     \
-        WarnL << ex.what();                                                    \
-        return;                                                                \
-    }
-#endif
-
-#ifndef MAX
-#define MAX(a, b) ((a) > (b) ? (a) : (b))
-#endif // MAX
-
-#ifndef MIN
-#define MIN(a, b) ((a) < (b) ? (a) : (b))
-#endif // MIN
-
-#ifndef CLEAR_ARR
-#define CLEAR_ARR(arr)                                                                                                 \
-    for (auto &item : arr) {                                                                                           \
-        item = 0;                                                                                                      \
-    }
-#endif // CLEAR_ARR
-
-#define RTSP_SCHEMA "rtsp"
-#define RTMP_SCHEMA "rtmp"
-#define TS_SCHEMA "ts"
-#define FMP4_SCHEMA "fmp4"
-#define HLS_SCHEMA "hls"
-#define HLS_FMP4_SCHEMA "hls.fmp4"
-
-#define VHOST_KEY "vhost"
-#define DEFAULT_VHOST "__defaultVhost__"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-extern void Assert_Throw(int failed, const char *exp, const char *func, const char *file, int line, const char *str);
-#ifdef __cplusplus
-}
-#endif
-
-namespace mediakit {
-
-class AssertFailedException : public std::runtime_error {
-public:
-    template<typename ...T>
-    AssertFailedException(T && ...args) : std::runtime_error(std::forward<T>(args)...) {}
-};
-
-extern const char kServerName[];
-
-template <typename... ARGS>
-void Assert_ThrowCpp(int failed, const char *exp, const char *func, const char *file, int line, ARGS &&...args) {
-    if (failed) {
-        std::stringstream ss;
-        toolkit::LoggerWrapper::appendLog(ss, std::forward<ARGS>(args)...);
-        Assert_Throw(failed, exp, func, file, line, ss.str().data());
-    }
-}
-
-} // namespace mediakit
-#endif // ZLMEDIAKIT_MACROS_H

webrtc_player/android/zlm/src/main/cpp/include/Common/strCoding.h

@@ -1,36 +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 SRC_HTTP_STRCODING_H_
-#define SRC_HTTP_STRCODING_H_
-
-#include <iostream>
-#include <string>
-#include <cstdint>
-namespace mediakit {
-
-class strCoding {
-public:
-    static std::string UrlEncodePath(const std::string &str); //url路径 utf8编码
-    static std::string UrlEncodeComponent(const std::string &str); // url参数 utf8编码
-    static std::string UrlDecodePath(const std::string &str); //url路径 utf8解码
-    static std::string UrlDecodeComponent(const std::string &str); // url参数 utf8解码
-#if defined(_WIN32)
-    static std::string UTF8ToGB2312(const std::string &str);//utf_8转为gb2312
-    static std::string GB2312ToUTF8(const std::string &str); //gb2312 转utf_8
-#endif//defined(_WIN32)
-private:
-    strCoding(void);
-    virtual ~strCoding(void);
-};
-
-} /* namespace mediakit */
-
-#endif /* SRC_HTTP_STRCODING_H_ */

webrtc_player/android/zlm/src/main/cpp/include/Extension/CommonRtmp.h

@@ -1,57 +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_COMMONRTMP_H
-#define ZLMEDIAKIT_COMMONRTMP_H
-
-#include "Frame.h"
-#include "Rtmp/RtmpCodec.h"
-
-namespace mediakit{
-
-/**
- * 通用 rtmp解码类
- */
-class CommonRtmpDecoder : public RtmpCodec {
-public:
-    using Ptr = std::shared_ptr<CommonRtmpDecoder>;
-
-    /**
-     * 构造函数
-     */
-    CommonRtmpDecoder(const Track::Ptr &track) : RtmpCodec(track) {}
-
-    /**
-     * 输入Rtmp并解码
-     * @param rtmp Rtmp数据包
-     */
-    void inputRtmp(const RtmpPacket::Ptr &rtmp) override;
-};
-
-/**
- * 通用 rtmp编码类
- */
-class CommonRtmpEncoder : public RtmpCodec {
-public:
-    using Ptr = std::shared_ptr<CommonRtmpEncoder>;
-
-    CommonRtmpEncoder(const Track::Ptr &track) : RtmpCodec(track) {}
-
-    /**
-     * 输入帧数据
-     */
-    bool inputFrame(const Frame::Ptr &frame) override;
-
-private:
-    uint8_t _audio_flv_flags { 0 };
-};
-
-}//namespace mediakit
-#endif //ZLMEDIAKIT_COMMONRTMP_H

webrtc_player/android/zlm/src/main/cpp/include/Extension/CommonRtp.h

@@ -1,66 +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_COMMONRTP_H
-#define ZLMEDIAKIT_COMMONRTP_H
-
-#include "Frame.h"
-#include "Rtsp/RtpCodec.h"
-
-namespace mediakit{
-
-/**
- * 通用 rtp解码类
- */
-class CommonRtpDecoder : public RtpCodec {
-public:
-    using Ptr = std::shared_ptr <CommonRtpDecoder>;
-
-    /**
-     * 构造函数
-     * @param codec 编码id
-     * @param max_frame_size 允许的最大帧大小
-     */
-    CommonRtpDecoder(CodecId codec, size_t max_frame_size = 2 * 1024);
-
-    /**
-     * 输入rtp并解码
-     * @param rtp rtp数据包
-     * @param key_pos 此参数内部强制转换为false,请忽略之
-     */
-    bool inputRtp(const RtpPacket::Ptr &rtp, bool key_pos = false) override;
-
-private:
-    void obtainFrame();
-
-private:
-    bool _drop_flag = false;
-    uint16_t _last_seq = 0;
-    uint64_t _last_stamp = 0;
-    size_t _max_frame_size;
-    CodecId _codec;
-    FrameImp::Ptr _frame;
-};
-
-/**
- * 通用 rtp编码类
- */
-class CommonRtpEncoder : public RtpCodec {
-public:
-    using Ptr = std::shared_ptr <CommonRtpEncoder>;
-
-    /**
-     * 输入帧数据并编码成rtp
-     */
-    bool inputFrame(const Frame::Ptr &frame) override;
-};
-
-}//namespace mediakit
-#endif //ZLMEDIAKIT_COMMONRTP_H

webrtc_player/android/zlm/src/main/cpp/include/Extension/Factory.h

@@ -1,120 +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_FACTORY_H
-#define ZLMEDIAKIT_FACTORY_H
-
-#include <string>
-#include "Rtmp/amf.h"
-#include "Extension/Track.h"
-#include "Extension/Frame.h"
-#include "Rtsp/RtpCodec.h"
-#include "Rtmp/RtmpCodec.h"
-#include "Util/onceToken.h"
-
-#define REGISTER_STATIC_VAR_INNER(var_name, line) var_name##_##line##__
-#define REGISTER_STATIC_VAR(var_name, line) REGISTER_STATIC_VAR_INNER(var_name, line)
-
-#define REGISTER_CODEC(plugin) \
-static toolkit::onceToken REGISTER_STATIC_VAR(s_token, __LINE__) ([]() { \
-    Factory::registerPlugin(plugin); \
-});
-
-namespace mediakit {
-
-struct CodecPlugin {
-    CodecId (*getCodec)();
-    Track::Ptr (*getTrackByCodecId)(int sample_rate, int channels, int sample_bit);
-    Track::Ptr (*getTrackBySdp)(const SdpTrack::Ptr &track);
-    RtpCodec::Ptr (*getRtpEncoderByCodecId)(uint8_t pt);
-    RtpCodec::Ptr (*getRtpDecoderByCodecId)();
-    RtmpCodec::Ptr (*getRtmpEncoderByTrack)(const Track::Ptr &track);
-    RtmpCodec::Ptr (*getRtmpDecoderByTrack)(const Track::Ptr &track);
-    Frame::Ptr (*getFrameFromPtr)(const char *data, size_t bytes, uint64_t dts, uint64_t pts);
-};
-
-class Factory {
-public:
-    /**
-     * 注册插件，非线程安全的
-     */
-    static void registerPlugin(const CodecPlugin &plugin);
-
-    /**
-     * 根据codec_id 获取track
-     * @param codecId 编码id
-     * @param sample_rate 采样率，视频固定为90000
-     * @param channels 音频通道数
-     * @param sample_bit 音频采样位数
-     */
-    static Track::Ptr getTrackByCodecId(CodecId codecId, int sample_rate = 0, int channels = 0, int sample_bit = 0);
-
-    ////////////////////////////////rtsp相关//////////////////////////////////
-    /**
-     * 根据sdp生成Track对象
-     */
-    static Track::Ptr getTrackBySdp(const SdpTrack::Ptr &track);
-
-    /**
-     * 根据c api 抽象的Track生成具体Track对象
-     */
-    static Track::Ptr getTrackByAbstractTrack(const Track::Ptr& track);
-
-    /**
-     * 根据codec id生成rtp编码器
-     * @param codec_id 编码id
-     * @param pt rtp payload type
-     */
-    static RtpCodec::Ptr getRtpEncoderByCodecId(CodecId codec_id, uint8_t pt);
-
-    /**
-     * 根据Track生成Rtp解包器
-     */
-    static RtpCodec::Ptr getRtpDecoderByCodecId(CodecId codec);
-
-
-    ////////////////////////////////rtmp相关//////////////////////////////////
-
-    /**
-     * 根据amf对象获取视频相应的Track
-     * @param amf rtmp metadata中的videocodecid的值
-     */
-    static Track::Ptr getVideoTrackByAmf(const AMFValue &amf);
-
-    /**
-     * 根据amf对象获取音频相应的Track
-     * @param amf rtmp metadata中的audiocodecid的值
-     */
-    static Track::Ptr getAudioTrackByAmf(const AMFValue& amf, int sample_rate, int channels, int sample_bit);
-
-    /**
-     * 根据Track获取Rtmp的编码器
-     * @param track 媒体描述对象
-     */
-    static RtmpCodec::Ptr getRtmpEncoderByTrack(const Track::Ptr &track);
-
-    /**
-     * 根据Track获取Rtmp的解码器
-     * @param track 媒体描述对象
-     */
-    static RtmpCodec::Ptr getRtmpDecoderByTrack(const Track::Ptr &track);
-
-    /**
-     * 根据codecId获取rtmp的codec描述
-     */
-    static AMFValue getAmfByCodecId(CodecId codecId);
-
-    static Frame::Ptr getFrameFromPtr(CodecId codec, const char *data, size_t size, uint64_t dts, uint64_t pts);
-    static Frame::Ptr getFrameFromBuffer(CodecId codec, toolkit::Buffer::Ptr data, uint64_t dts, uint64_t pts);
-};
-
-}//namespace mediakit
-
-#endif //ZLMEDIAKIT_FACTORY_H