From 2927a53d9ad3d7cba61e3642fa789de0dd9a2542 Mon Sep 17 00:00:00 2001
From: Harsh Patel <babulpatel1309@gmail.com>
Date: Wed, 19 May 2021 13:51:18 +0530
Subject: [PATCH] - Updated flutter version and null safety

---
 lib/flutter_live.dart | 38 +++++++++++++++++++-------------------
 pubspec.yaml          |  8 ++++++--
 2 files changed, 25 insertions(+), 21 deletions(-)

diff --git a/lib/flutter_live.dart b/lib/flutter_live.dart
index 6cd7607..18f52a6 100644
--- a/lib/flutter_live.dart
+++ b/lib/flutter_live.dart
@@ -110,9 +110,9 @@ class RealtimePlayer {
 ///   streamUrl: "webrtc://d.ossrs.net:11985/live/livestream"
 class WebRTCUri {
   /// The api server url for WebRTC streaming.
-  String api;
+  String? api;
   /// The stream url to play or publish.
-  String streamUrl;
+  String? streamUrl;
 
   /// Parse the url to WebRTC uri.
   static WebRTCUri parse(String url) {
@@ -120,7 +120,7 @@ class WebRTCUri {
 
     var schema = 'https'; // For native, default to HTTPS
     if (uri.queryParameters.containsKey('schema')) {
-      schema = uri.queryParameters['schema'];
+      schema = uri.queryParameters['schema']!;
     } else {
       schema = 'https';
     }
@@ -134,7 +134,7 @@ class WebRTCUri {
 
     var api = '/rtc/v1/play/';
     if (uri.queryParameters.containsKey('play')) {
-      api = uri.queryParameters['play'];
+      api = uri.queryParameters['play']!;
     }
 
     var apiParams = [];
@@ -159,8 +159,8 @@ class WebRTCUri {
 
 /// A WebRTC player, using [flutter_webrtc](https://pub.dev/packages/flutter_webrtc)
 class WebRTCPlayer {
-  webrtc.AddStreamCallback _onRemoteStream;
-  webrtc.RTCPeerConnection _pc;
+  webrtc.AddStreamCallback? _onRemoteStream;
+  webrtc.RTCPeerConnection? _pc;
 
   /// When got a remote stream.
   set onRemoteStream(webrtc.AddStreamCallback v) {
@@ -175,7 +175,7 @@ class WebRTCPlayer {
   /// [url] must a path parsed by [WebRTCUri.parse] in https://github.com/rtcdn/rtcdn-draft
   Future<void> play(String url) async {
     if (_pc != null) {
-      await _pc.close();
+      await _pc!.close();
     }
 
     // Create the peer connection.
@@ -187,36 +187,36 @@ class WebRTCPlayer {
     print('WebRTC: createPeerConnection done');
 
     // Setup the peer connection.
-    _pc.onAddStream = (webrtc.MediaStream stream) {
+    _pc!.onAddStream = (webrtc.MediaStream stream) {
       print('WebRTC: got stream ${stream.id}');
       if (_onRemoteStream == null) {
         print('Warning: Stream ${stream.id} is leak');
         return;
       }
-      _onRemoteStream(stream);
+      _onRemoteStream!(stream);
     };
 
-    _pc.addTransceiver(
+    _pc!.addTransceiver(
         kind: webrtc.RTCRtpMediaType.RTCRtpMediaTypeAudio,
         init: webrtc.RTCRtpTransceiverInit(direction: webrtc.TransceiverDirection.RecvOnly),
     );
-    _pc.addTransceiver(
+    _pc!.addTransceiver(
       kind: webrtc.RTCRtpMediaType.RTCRtpMediaTypeVideo,
       init: webrtc.RTCRtpTransceiverInit(direction: webrtc.TransceiverDirection.RecvOnly),
     );
     print('WebRTC: Setup PC done, A|V RecvOnly');
 
     // Start SDP handshake.
-    webrtc.RTCSessionDescription offer = await _pc.createOffer({
+    webrtc.RTCSessionDescription offer = await _pc!.createOffer({
       'mandatory': {'OfferToReceiveAudio': true, 'OfferToReceiveVideo': true},
     });
-    await _pc.setLocalDescription(offer);
-    print('WebRTC: createOffer, ${offer.type} is ${offer.sdp.replaceAll('\n', '\\n').replaceAll('\r', '\\r')}');
+    await _pc!.setLocalDescription(offer);
+    print('WebRTC: createOffer, ${offer.type} is ${offer.sdp!.replaceAll('\n', '\\n').replaceAll('\r', '\\r')}');
 
-    webrtc.RTCSessionDescription answer = await _handshake(url, offer.sdp);
-    print('WebRTC: got ${answer.type} is ${answer.sdp.replaceAll('\n', '\\n').replaceAll('\r', '\\r')}');
+    webrtc.RTCSessionDescription answer = await _handshake(url, offer.sdp!);
+    print('WebRTC: got ${answer.type} is ${answer.sdp!.replaceAll('\n', '\\n').replaceAll('\r', '\\r')}');
 
-    await _pc.setRemoteDescription(answer);
+    await _pc!.setRemoteDescription(answer);
   }
 
   /// Handshake to exchange SDP, send offer and got answer.
@@ -238,7 +238,7 @@ class WebRTCPlayer {
       //    {api: "xxx", sdp: "offer", streamurl: "webrtc://d.ossrs.net:11985/live/livestream"}
       // Response:
       //    {code: 0, sdp: "answer", sessionid: "007r51l7:X2Lv"}
-      HttpClientRequest req = await client.postUrl(Uri.parse(uri.api));
+      HttpClientRequest req = await client.postUrl(Uri.parse(uri.api!));
       req.headers.set('Content-Type', 'application/json');
       req.add(utf8.encode(json.encode({'api': uri.api, 'streamurl': uri.streamUrl, 'sdp': offer})));
       print('WebRTC request: ${uri.api} offer=${offer.length}B');
@@ -261,7 +261,7 @@ class WebRTCPlayer {
   /// Dispose the player.
   void dispose() {
     if (_pc != null) {
-      _pc.close();
+      _pc!.close();
     }
   }
 }
diff --git a/pubspec.yaml b/pubspec.yaml
index a872e2f..c80f916 100644
--- a/pubspec.yaml
+++ b/pubspec.yaml
@@ -4,7 +4,7 @@ description: Live streaming player, iOS+Android, RTMP/HTTP-FLV/HLS/WebRTC, by Fl
 homepage: https://github.com/ossrs/flutter_live
 
 environment:
-  sdk: ">=2.7.0 <3.0.0"
+  sdk: ">=2.12.0 <3.0.0"
   flutter: ">=1.20.0 <2.0.0"
 
 dependencies:
@@ -12,7 +12,11 @@ dependencies:
     sdk: flutter
   fijkplayer: ^0.8.8
   flutter_webrtc: ^0.6.3
-  camera_with_rtmp: ^0.3.2
+  camera_with_rtmp:
+    git:
+      url: https://github.com/babulpatel1309/flutter_rtmppublisher.git
+      ref: 963c0b82cd94f09f1e4387d33a89e3b067957c19
+
 
 dev_dependencies:
   flutter_test: